From 016fcab8ff46fca29375d484226ec91932aa4a07 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 4 Jul 2013 20:01:02 -0300 Subject: ASoC: sglt5000: Fix the default value of CHIP_SSS_CTRL According to the sgtl5000 reference manual, the default value of CHIP_SSS_CTRL is 0x10. Reported-by: Oskar Schirmer Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559dc92c..d659d3adcfb3 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -38,7 +38,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, - { SGTL5000_CHIP_SSS_CTRL, 0x0008 }, + { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, -- cgit v1.2.3-70-g09d2 From 5c78dfe87ea04b501ee000a7f03b9432ac9d008c Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 4 Jul 2013 20:01:03 -0300 Subject: ASoC: sglt5000: Fix SGTL5000_PLL_FRAC_DIV_MASK SGTL5000_PLL_FRAC_DIV_MASK is used to mask bits 0-10 (11 bits in total) of register CHIP_PLL_CTRL, so fix the mask to accomodate all this bit range. Reported-by: Oskar Schirmer Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sgtl5000.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 4b69229a9818..2f8c88931f69 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -347,7 +347,7 @@ #define SGTL5000_PLL_INT_DIV_MASK 0xf800 #define SGTL5000_PLL_INT_DIV_SHIFT 11 #define SGTL5000_PLL_INT_DIV_WIDTH 5 -#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x07ff #define SGTL5000_PLL_FRAC_DIV_SHIFT 0 #define SGTL5000_PLL_FRAC_DIV_WIDTH 11 -- cgit v1.2.3-70-g09d2 From 82e414fa1dbbc07e7b6d582e4fbcc9b0a5299f7f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 5 Jul 2013 12:36:24 +0100 Subject: ASoC: wm8994: Remove overly noisy debug logging This was committed in error. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 25580b5a853f..1b89aa9029e8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3856,8 +3856,6 @@ static void wm8958_mic_work(struct work_struct *work) mic_complete_work.work); struct snd_soc_codec *codec = wm8994->hubs.codec; - dev_crit(codec->dev, "MIC WORK %x\n", wm8994->mic_status); - pm_runtime_get_sync(codec->dev); mutex_lock(&wm8994->accdet_lock); @@ -3867,8 +3865,6 @@ static void wm8958_mic_work(struct work_struct *work) mutex_unlock(&wm8994->accdet_lock); pm_runtime_put(codec->dev); - - dev_crit(codec->dev, "MIC WORK %x DONE\n", wm8994->mic_status); } static irqreturn_t wm8958_mic_irq(int irq, void *data) -- cgit v1.2.3-70-g09d2 From 770100108be7dbe614361dbcc450096b4cdfc98b Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Thu, 11 Jul 2013 12:38:25 +0530 Subject: ASoC: Samsung: Set RFS and BFS in slave mode As per the User Manual, the RFS and BFS should be set in slave mode for correct operation. Signed-off-by: Padmavathi Venna Signed-off-by: Andrew Bresticker Reviewed-by: Simon Glass Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 7a1734697434..959c702235c8 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -742,13 +742,13 @@ static int config_setup(struct i2s_dai *i2s) return -EAGAIN; } - /* Don't bother RFS, BFS & PSR in Slave mode */ - if (is_slave(i2s)) - return 0; - set_bfs(i2s, bfs); set_rfs(i2s, rfs); + /* Don't bother with PSR in Slave mode */ + if (is_slave(i2s)) + return 0; + if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { psr = i2s->rclk_srcrate / i2s->frmclk / rfs; writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR); -- cgit v1.2.3-70-g09d2 From f6becf0b2ffef0bed813d7f910b5d276c5dc45e1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 11 Jul 2013 14:35:43 +0200 Subject: ASoC: omap-pcm: Request the DMA channel differently when DT is involved When booting with DT the platform_get_resource_byname() is not available to get the DMA resource. In this case the DAI drivers will set the filter_data to the name of the DMA and omap-pcm can use this to request the DMA channel. Signed-off-by: Peter Ujfalusi Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index c28e042f2208..a11405de86e8 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -113,14 +113,25 @@ static int omap_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_dmaengine_dai_dma_data *dma_data; + int ret; snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware); dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - return snd_dmaengine_pcm_open_request_chan(substream, - omap_dma_filter_fn, - dma_data->filter_data); + /* DT boot: filter_data is the DMA name */ + if (rtd->cpu_dai->dev->of_node) { + struct dma_chan *chan; + + chan = dma_request_slave_channel(rtd->cpu_dai->dev, + dma_data->filter_data); + ret = snd_dmaengine_pcm_open(substream, chan); + } else { + ret = snd_dmaengine_pcm_open_request_chan(substream, + omap_dma_filter_fn, + dma_data->filter_data); + } + return ret; } static int omap_pcm_mmap(struct snd_pcm_substream *substream, -- cgit v1.2.3-70-g09d2 From a8035f073cb508a0c1223db2662510575627b41d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 11 Jul 2013 14:35:44 +0200 Subject: ASoC: omap-mcpdm: Do not use platform_get_resource_byname() for DMA The DMA resource no longer available via this API when booting with DT. McPDM is only available on OMAP4/5 and both can boot with DT only. Set the dma_data.filter_data to the DMA name which will be used by omap-pcm to request the DMA channel. Signed-off-by: Peter Ujfalusi Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index eb05c7ed6d05..a49dc52f8abc 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -66,7 +66,6 @@ struct omap_mcpdm { bool restart; struct snd_dmaengine_dai_dma_data dma_data[2]; - unsigned int dma_req[2]; }; /* @@ -477,19 +476,8 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dma_data[0].addr = res->start + MCPDM_REG_DN_DATA; mcpdm->dma_data[1].addr = res->start + MCPDM_REG_UP_DATA; - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "dn_link"); - if (!res) - return -ENODEV; - - mcpdm->dma_req[0] = res->start; - mcpdm->dma_data[0].filter_data = &mcpdm->dma_req[0]; - - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "up_link"); - if (!res) - return -ENODEV; - - mcpdm->dma_req[1] = res->start; - mcpdm->dma_data[1].filter_data = &mcpdm->dma_req[1]; + mcpdm->dma_data[0].filter_data = "dn_link"; + mcpdm->dma_data[1].filter_data = "up_link"; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (res == NULL) -- cgit v1.2.3-70-g09d2 From 2ebef44789223389708505e33c67d44e9f999d4a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 11 Jul 2013 14:35:45 +0200 Subject: ASoC: omap-dmic: Do not use platform_get_resource_byname() for DMA The DMA resource no longer available via this API when booting with DT. DMIC is only available on OMAP4/5 and both can boot with DT only. Set the dma_data.filter_data to the DMA name which will be used by omap-pcm to request the DMA channel. Signed-off-by: Peter Ujfalusi Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 2ad0370146fd..4db1f8e6e172 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -57,7 +57,6 @@ struct omap_dmic { struct mutex mutex; struct snd_dmaengine_dai_dma_data dma_data; - unsigned int dma_req; }; static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val) @@ -478,15 +477,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) } dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG; - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(dmic->dev, "invalid dma resource\n"); - ret = -ENODEV; - goto err_put_clk; - } - - dmic->dma_req = res->start; - dmic->dma_data.filter_data = &dmic->dma_req; + dmic->dma_data.filter_data = "up_link"; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!res) { -- cgit v1.2.3-70-g09d2 From 9ab1fac4829b3da0ba4d3f44d95d3e8ad13e6629 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 11 Jul 2013 14:35:46 +0200 Subject: ASoC: omap-mcbsp: Use different method for DMA request when booted with DT The DMA resource no longer available via this API when booting with DT. When the board is booted with DT do not use platform_get_resource_byname(), instead set the dma_data.filter_data to the name of the DMA channel and omap-pcm can use this name to request the DMA channel. Signed-off-by: Peter Ujfalusi Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/mcbsp.c | 39 ++++++++++++++++++++++----------------- 1 file changed, 22 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index eb68c7db1cf3..361e4c03646e 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -1012,28 +1012,33 @@ int omap_mcbsp_init(struct platform_device *pdev) } } - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); - if (!res) { - dev_err(&pdev->dev, "invalid rx DMA channel\n"); - return -ENODEV; - } - /* RX DMA request number, and port address configuration */ - mcbsp->dma_req[1] = res->start; - mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; - mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, 1); - mcbsp->dma_data[1].maxburst = 4; + if (!pdev->dev.of_node) { + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); + if (!res) { + dev_err(&pdev->dev, "invalid tx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[0] = res->start; + mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); - if (!res) { - dev_err(&pdev->dev, "invalid tx DMA channel\n"); - return -ENODEV; + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); + if (!res) { + dev_err(&pdev->dev, "invalid rx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[1] = res->start; + mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; + } else { + mcbsp->dma_data[0].filter_data = "tx"; + mcbsp->dma_data[1].filter_data = "rx"; } - /* TX DMA request number, and port address configuration */ - mcbsp->dma_req[0] = res->start; - mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; + mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, 0); mcbsp->dma_data[0].maxburst = 4; + mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, 1); + mcbsp->dma_data[1].maxburst = 4; + mcbsp->fclk = clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { ret = PTR_ERR(mcbsp->fclk); -- cgit v1.2.3-70-g09d2 From 5f17482a3244c07646279d16c0e5b8c0b2b76d0e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 11 Jul 2013 20:09:43 -0700 Subject: ASoC: wm8978: enable symmetric rates wm8978 needs .symmetric_rates = 1. The playback/capture will be strange without this patch when it used asymmetric rate in same time Tested-by: Yusuke Goda Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 029f31c8e703..d8fc531c0e59 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -921,6 +921,7 @@ static struct snd_soc_dai_driver wm8978_dai = { .formats = WM8978_FORMATS, }, .ops = &wm8978_dai_ops, + .symmetric_rates = 1, }; static int wm8978_suspend(struct snd_soc_codec *codec) -- cgit v1.2.3-70-g09d2 From e394fe55f7cf5a4f6c20fbd02ab37b1d5c3dd364 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 1 Jul 2013 16:49:04 +0800 Subject: ASoC: adav80x: Add module device table for adav801 This driver can be built as module, thus add module device table for adav801 to support module auto loading. To make the naming consistent, also rename adav80x_id to adav80x_i2c_id. Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 3c839cc4e00e..15b012d0f226 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -868,6 +868,12 @@ static int adav80x_bus_remove(struct device *dev) } #if defined(CONFIG_SPI_MASTER) +static const struct spi_device_id adav80x_spi_id[] = { + { "adav801", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adav80x_spi_id); + static int adav80x_spi_probe(struct spi_device *spi) { return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); @@ -885,15 +891,16 @@ static struct spi_driver adav80x_spi_driver = { }, .probe = adav80x_spi_probe, .remove = adav80x_spi_remove, + .id_table = adav80x_spi_id, }; #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static const struct i2c_device_id adav80x_id[] = { +static const struct i2c_device_id adav80x_i2c_id[] = { { "adav803", 0 }, { } }; -MODULE_DEVICE_TABLE(i2c, adav80x_id); +MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) @@ -913,7 +920,7 @@ static struct i2c_driver adav80x_i2c_driver = { }, .probe = adav80x_i2c_probe, .remove = adav80x_i2c_remove, - .id_table = adav80x_id, + .id_table = adav80x_i2c_id, }; #endif -- cgit v1.2.3-70-g09d2 From a2911cdb1fd09de7c0da3938ffab1bc5cedda4f9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 3 Jul 2013 21:15:13 -0700 Subject: ASoC: add ak4554 driver ak4554 is very simple DA/AD converter which has no setting register. Note that it has hard coded asymmetric data format playback : SND_SOC_DAIFMT_RIGHT_J capture : SND_SOC_DAIFMT_LEFT_J This driver has single DAI and doesn't have set_fmt. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 ++ sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/ak4554.c | 79 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 84 insertions(+) create mode 100644 sound/soc/codecs/ak4554.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..ffb9adb18647 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -198,6 +198,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4554 + tristate + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd8066f546..fab4086796a0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4554-objs := ak4554.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -138,6 +139,7 @@ obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c new file mode 100644 index 000000000000..c1a1733f8a35 --- /dev/null +++ b/sound/soc/codecs/ak4554.c @@ -0,0 +1,79 @@ +/* + * ak4554.c + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include + +/* + * ak4554 is very simple DA/AD converter which has no setting register. + * + * CAUTION + * + * ak4554 playback format is SND_SOC_DAIFMT_RIGHT_J, + * and, capture format is SND_SOC_DAIFMT_LEFT_J + * on same bit clock, LR clock. + * But, this driver doesn't have snd_soc_dai_ops :: set_fmt + * + * CPU/Codec DAI image + * + * CPU-DAI1 (plaback only fmt = RIGHT_J) --+-- ak4554 + * | + * CPU-DAI2 (capture only fmt = LEFT_J) ---+ + */ + +static struct snd_soc_dai_driver ak4554_dai = { + .name = "ak4554-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { +}; + +static int ak4554_soc_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_ak4554, + &ak4554_dai, 1); +} + +static int ak4554_soc_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ak4554_driver = { + .driver = { + .name = "ak4554-adc-dac", + .owner = THIS_MODULE, + }, + .probe = ak4554_soc_probe, + .remove = ak4554_soc_remove, +}; +module_platform_driver(ak4554_driver); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SoC AK4554 driver"); +MODULE_AUTHOR("Kuninori Morimoto "); -- cgit v1.2.3-70-g09d2 From b25f77815021ec6e7400a82c4984b9c80699ce80 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 4 Jul 2013 19:42:49 -0700 Subject: ASoC: ak4554: add DT support Support for loading the ak4554 codec module via devicetree. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/ak4554.c | 11 +++++++++++ sound/soc/codecs/ak4554.c | 7 +++++++ 2 files changed, 18 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4554.c (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/ak4554.c b/Documentation/devicetree/bindings/sound/ak4554.c new file mode 100644 index 000000000000..934fa02754b3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4554.c @@ -0,0 +1,11 @@ +AK4554 ADC/DAC + +Required properties: + + - compatible : "asahi-kasei,ak4554" + +Example: + +ak4554-adc-dac { + compatible = "asahi-kasei,ak4554"; +}; diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index c1a1733f8a35..6aed9c4d06d6 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -64,10 +64,17 @@ static int ak4554_soc_remove(struct platform_device *pdev) return 0; } +static struct of_device_id ak4554_of_match[] = { + { .compatible = "asahi-kasei,ak4554" }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4554_of_match); + static struct platform_driver ak4554_driver = { .driver = { .name = "ak4554-adc-dac", .owner = THIS_MODULE, + .of_match_table = ak4554_of_match, }, .probe = ak4554_soc_probe, .remove = ak4554_soc_remove, -- cgit v1.2.3-70-g09d2 From b63144e6c6097486e7678f9ecc2769b68d2ec83e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 4 Jul 2013 08:56:28 +0100 Subject: ASoC: arizona: Add signal activity output for DRC When doing signal activity detection, the only output from the DRC will often be a GPIO pin. This patch adds a signal activity output that is activated when a GPIO is configured to output the DRC signal activity detection. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5102.c | 6 ++++++ sound/soc/codecs/wm5110.c | 9 +++++++++ 4 files changed, 49 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index de625813c0e6..c9116ac0d4a7 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -19,6 +19,7 @@ #include #include +#include #include #include "arizona.h" @@ -223,6 +224,38 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +int arizona_init_gpio(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int i; + + switch (arizona->type) { + case WM5110: + snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity"); + } + + snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity"); + + for (i = 0; i < ARRAY_SIZE(arizona->pdata.gpio_defaults); i++) { + switch (arizona->pdata.gpio_defaults[i] & ARIZONA_GPN_FN_MASK) { + case ARIZONA_GP_FN_DRC1_SIGNAL_DETECT: + snd_soc_dapm_enable_pin(&codec->dapm, + "DRC1 Signal Activity"); + break; + case ARIZONA_GP_FN_DRC2_SIGNAL_DETECT: + snd_soc_dapm_enable_pin(&codec->dapm, + "DRC2 Signal Activity"); + break; + default: + break; + } + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_gpio); + const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "None", "Tone Generator 1", diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b60b08ccc1d0..fe1b794bd5f0 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -242,6 +242,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout); extern int arizona_init_spk(struct snd_soc_codec *codec); +extern int arizona_init_gpio(struct snd_soc_codec *codec); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 282fd232cdf7..a6cbdb4b5c0f 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -998,6 +998,8 @@ SND_SOC_DAPM_INPUT("IN2R"), SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), + SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | @@ -1614,6 +1616,9 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "SPKDAT1R", NULL, "OUT5R" }, { "MICSUPP", NULL, "SYSCLK" }, + + { "DRC1 Signal Activity", NULL, "DRC1L" }, + { "DRC1 Signal Activity", NULL, "DRC1R" }, }; static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, @@ -1781,6 +1786,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) return ret; arizona_init_spk(codec); + arizona_init_gpio(codec); snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e7cb4ba161a..fc410377277f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -432,6 +432,9 @@ SND_SOC_DAPM_INPUT("IN3R"), SND_SOC_DAPM_INPUT("IN4L"), SND_SOC_DAPM_INPUT("IN4R"), +SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), +SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"), + SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | @@ -1006,6 +1009,11 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "SPKDAT2R", NULL, "OUT6R" }, { "MICSUPP", NULL, "SYSCLK" }, + + { "DRC1 Signal Activity", NULL, "DRC1L" }, + { "DRC1 Signal Activity", NULL, "DRC1R" }, + { "DRC2 Signal Activity", NULL, "DRC2L" }, + { "DRC2 Signal Activity", NULL, "DRC2R" }, }; static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, @@ -1170,6 +1178,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) return ret; arizona_init_spk(codec); + arizona_init_gpio(codec); snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); -- cgit v1.2.3-70-g09d2 From b79fae606c921522577f3000b6b9a807cd733d2e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 4 Jul 2013 16:53:03 +0100 Subject: ASoC: arizona: Add default case to silence build warning Reported-by: Fengguang Wu Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index c9116ac0d4a7..8dc6881496de 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -233,6 +233,9 @@ int arizona_init_gpio(struct snd_soc_codec *codec) switch (arizona->type) { case WM5110: snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity"); + break; + default: + break; } snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity"); -- cgit v1.2.3-70-g09d2 From 01f00d55a7f21b966417fece78214154f01590ed Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 3 Jul 2013 16:37:56 +0800 Subject: ASoC: atmel_ssc_dai: move set dma data to startup callback move set dma data to startup callback function, if the set dma data exist in hw_params callback, so the dma data only usable when call hw_params, if want use it before hw_params callback, it will cause NULL pointer access oops Signed-off-by: Bo Shen Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 33 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f3fdfa07fcb9..6cf9cf1238cc 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -196,15 +196,27 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; - int dir_mask; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dir = 0; dir_mask = SSC_DIR_MASK_PLAYBACK; - else + } else { + dir = 1; dir_mask = SSC_DIR_MASK_CAPTURE; + } + + dma_params = &ssc_dma_params[dai->id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + snd_soc_dai_set_dma_data(dai, substream, dma_params); spin_lock_irq(&ssc_p->lock); if (ssc_p->dir_mask & dir_mask) { @@ -325,7 +337,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); int id = dai->id; struct atmel_ssc_info *ssc_p = &ssc_info[id]; struct atmel_pcm_dma_params *dma_params; @@ -344,19 +355,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, else dir = 1; - dma_params = &ssc_dma_params[id][dir]; - dma_params->ssc = ssc_p->ssc; - dma_params->substream = substream; - - ssc_p->dma_params[dir] = dma_params; - - /* - * The snd_soc_pcm_stream->dma_data field is only used to communicate - * the appropriate DMA parameters to the pcm driver hw_params() - * function. It should not be used for other purposes - * as it is common to all substreams. - */ - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_params); + dma_params = ssc_p->dma_params[dir]; channels = params_channels(params); -- cgit v1.2.3-70-g09d2 From f1b0dd8b9377590b387fd21ba67081ed0e7111e3 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 3 Jul 2013 16:37:57 +0800 Subject: ASoC: atmel_ssc_dai: add error mask define add error mask define, which will be used when execute DMA transfer Signed-off-by: Bo Shen Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6cf9cf1238cc..1ab47639f11c 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -73,6 +73,7 @@ static struct atmel_ssc_mask ssc_tx_mask = { .ssc_disable = SSC_BIT(CR_TXDIS), .ssc_endx = SSC_BIT(SR_ENDTX), .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .ssc_error = SSC_BIT(SR_OVRUN), .pdc_enable = ATMEL_PDC_TXTEN, .pdc_disable = ATMEL_PDC_TXTDIS, }; @@ -82,6 +83,7 @@ static struct atmel_ssc_mask ssc_rx_mask = { .ssc_disable = SSC_BIT(CR_RXDIS), .ssc_endx = SSC_BIT(SR_ENDRX), .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .ssc_error = SSC_BIT(SR_OVRUN), .pdc_enable = ATMEL_PDC_RXTEN, .pdc_disable = ATMEL_PDC_RXTDIS, }; -- cgit v1.2.3-70-g09d2 From cede8d7aaa60bd7c03b9ec5eb43b09714710b8ba Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 3 Jul 2013 16:37:58 +0800 Subject: ASoC: atmel-pcm-dma: move prepare for dma to dai prepare as prepare callback for dma is acctually access ssc register which better done in dai driver, so move it to dai prepare callback function Signed-off-by: Bo Shen Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm-dma.c | 14 -------------- sound/soc/atmel/atmel_ssc_dai.c | 1 + 2 files changed, 1 insertion(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 1d38fd0bc4e2..5a57803cb180 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -175,19 +175,6 @@ err: return ret; } -static int atmel_pcm_dma_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_pcm_dma_params *prtd; - - prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - ssc_writex(prtd->ssc->regs, SSC_IER, prtd->mask->ssc_error); - ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_enable); - - return 0; -} - static int atmel_pcm_open(struct snd_pcm_substream *substream) { snd_soc_set_runtime_hwparams(substream, &atmel_pcm_dma_hardware); @@ -200,7 +187,6 @@ static struct snd_pcm_ops atmel_pcm_ops = { .close = snd_dmaengine_pcm_close_release_chan, .ioctl = snd_pcm_lib_ioctl, .hw_params = atmel_pcm_hw_params, - .prepare = atmel_pcm_dma_prepare, .trigger = snd_dmaengine_pcm_trigger, .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = atmel_pcm_mmap, diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 1ab47639f11c..0ecf356027f6 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -649,6 +649,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + ssc_writel(ssc_p->ssc->regs, IER, dma_params->mask->ssc_error); pr_debug("%s enabled SSC_SR=0x%08x\n", dir ? "receive" : "transmit", -- cgit v1.2.3-70-g09d2 From 8331b9e332a6e72d5285b05f56a7b66b692cb67a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Jul 2013 11:58:55 +0200 Subject: sound: oss/vwsnd: Add missing inclusion of linux/delay.h Reported-by: Fengguang Wu Signed-off-by: Takashi Iwai --- sound/oss/vwsnd.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 7e814a5c3677..d8db9023bc55 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -149,6 +149,7 @@ #include #include #include +#include #include -- cgit v1.2.3-70-g09d2 From 95e0e07e710e24a240f2c9645ecaad3559d6040d Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 3 Jul 2013 16:38:00 +0800 Subject: ASoC: atmel-pcm: use generic dmaengine framework Align atmel pcm to use ASoC generic dmaengine framework DMA is fully device tree based Signed-off-by: Bo Shen Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 1 + sound/soc/atmel/atmel-pcm-dma.c | 104 +++++++--------------------------------- 2 files changed, 17 insertions(+), 88 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 3fdd87fa18a9..1c0b1858638c 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -13,6 +13,7 @@ config SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_DMA tristate depends on SND_ATMEL_SOC + select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC tristate diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 5a57803cb180..3ff5601eef10 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -89,124 +89,52 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, } } -/*--------------------------------------------------------------------------*\ - * DMAENGINE operations -\*--------------------------------------------------------------------------*/ -static bool filter(struct dma_chan *chan, void *slave) -{ - struct at_dma_slave *sl = slave; - - if (sl->dma_dev == chan->device->dev) { - chan->private = sl; - return true; - } else { - return false; - } -} - static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, struct atmel_pcm_dma_params *prtd) + struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pcm_dma_params *prtd; struct ssc_device *ssc; - struct dma_chan *dma_chan; - struct dma_slave_config slave_config; int ret; + prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ssc = prtd->ssc; - ret = snd_hwparams_to_dma_slave_config(substream, params, - &slave_config); + ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); if (ret) { pr_err("atmel-pcm: hwparams to dma slave configure failed\n"); return ret; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = (dma_addr_t)ssc->phybase + SSC_THR; - slave_config.dst_maxburst = 1; + slave_config->dst_addr = ssc->phybase + SSC_THR; + slave_config->dst_maxburst = 1; } else { - slave_config.src_addr = (dma_addr_t)ssc->phybase + SSC_RHR; - slave_config.src_maxburst = 1; - } - - dma_chan = snd_dmaengine_pcm_get_chan(substream); - if (dmaengine_slave_config(dma_chan, &slave_config)) { - pr_err("atmel-pcm: failed to configure dma channel\n"); - ret = -EBUSY; - return ret; - } - - return 0; -} - -static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_pcm_dma_params *prtd; - struct ssc_device *ssc; - struct at_dma_slave *sdata = NULL; - int ret; - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - ssc = prtd->ssc; - if (ssc->pdev) - sdata = ssc->pdev->dev.platform_data; - - ret = snd_dmaengine_pcm_open_request_chan(substream, filter, sdata); - if (ret) { - pr_err("atmel-pcm: dmaengine pcm open failed\n"); - return -EINVAL; - } - - ret = atmel_pcm_configure_dma(substream, params, prtd); - if (ret) { - pr_err("atmel-pcm: failed to configure dmai\n"); - goto err; + slave_config->src_addr = ssc->phybase + SSC_RHR; + slave_config->src_maxburst = 1; } prtd->dma_intr_handler = atmel_pcm_dma_irq; - return 0; -err: - snd_dmaengine_pcm_close_release_chan(substream); - return ret; -} - -static int atmel_pcm_open(struct snd_pcm_substream *substream) -{ - snd_soc_set_runtime_hwparams(substream, &atmel_pcm_dma_hardware); - return 0; } -static struct snd_pcm_ops atmel_pcm_ops = { - .open = atmel_pcm_open, - .close = snd_dmaengine_pcm_close_release_chan, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = atmel_pcm_hw_params, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer_no_residue, - .mmap = atmel_pcm_mmap, -}; - -static struct snd_soc_platform_driver atmel_soc_platform = { - .ops = &atmel_pcm_ops, - .pcm_new = atmel_pcm_new, - .pcm_free = atmel_pcm_free, +static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = { + .prepare_slave_config = atmel_pcm_configure_dma, + .pcm_hardware = &atmel_pcm_dma_hardware, + .prealloc_buffer_size = ATMEL_SSC_DMABUF_SIZE, }; int atmel_pcm_dma_platform_register(struct device *dev) { - return snd_soc_register_platform(dev, &atmel_soc_platform); + return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); } EXPORT_SYMBOL(atmel_pcm_dma_platform_register); void atmel_pcm_dma_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(dev); + snd_dmaengine_pcm_unregister(dev); } EXPORT_SYMBOL(atmel_pcm_dma_platform_unregister); -- cgit v1.2.3-70-g09d2 From 4b8846062faac4e5c3f08e2e06bbb33c949aa51f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Jul 2013 12:00:24 +0200 Subject: sound: oss/vwsnd: Always define vwsnd_mutex While the conversion of BKL to mutex in commit 645ef9ef, the mutex definition was put in a wrong place inside #ifdef WSND_DEBUG, which leads to the build error. Just move it outside the ifdef. Reported-by: Fengguang Wu Signed-off-by: Takashi Iwai --- sound/oss/vwsnd.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index d8db9023bc55..4bbcc0fcd4eb 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -155,12 +155,13 @@ #include "sound_config.h" +static DEFINE_MUTEX(vwsnd_mutex); + /*****************************************************************************/ /* debug stuff */ #ifdef VWSND_DEBUG -static DEFINE_MUTEX(vwsnd_mutex); static int shut_up = 1; /* -- cgit v1.2.3-70-g09d2 From 35261edca31b0084bb615e6b50dfa5757b26db76 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 2 Jul 2013 17:19:25 +0800 Subject: ASoC: db1200: add .owner to struct snd_soc_card Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/au1x/db1200.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index a497a0cfeba1..decba87a074c 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -73,12 +73,14 @@ static struct snd_soc_dai_link db1300_ac97_dai = { static struct snd_soc_card db1300_ac97_machine = { .name = "DB1300_AC97", + .owner = THIS_MODULE, .dai_link = &db1300_ac97_dai, .num_links = 1, }; static struct snd_soc_card db1550_ac97_machine = { .name = "DB1550_AC97", + .owner = THIS_MODULE, .dai_link = &db1200_ac97_dai, .num_links = 1, }; @@ -145,6 +147,7 @@ static struct snd_soc_dai_link db1300_i2s_dai = { static struct snd_soc_card db1300_i2s_machine = { .name = "DB1300_I2S", + .owner = THIS_MODULE, .dai_link = &db1300_i2s_dai, .num_links = 1, }; @@ -161,6 +164,7 @@ static struct snd_soc_dai_link db1550_i2s_dai = { static struct snd_soc_card db1550_i2s_machine = { .name = "DB1550_I2S", + .owner = THIS_MODULE, .dai_link = &db1550_i2s_dai, .num_links = 1, }; -- cgit v1.2.3-70-g09d2 From 64b0c282e87948bc0c5a1b94ca3d4dd9f6415c6f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 25 Jun 2013 11:00:06 +0100 Subject: ASoC: codecs: Make ALL_CODECS depend on COMPILE_TEST The main function of the option is to enable compile testing. There is still an option since COMPILE_TEST is intended to enable selection of extra drivers rather than forcing them on. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..01d112b48e7e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" + depends on COMPILE_TEST select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE -- cgit v1.2.3-70-g09d2 From 7464dcd061045e664723acbf6be49e3ff53d97d8 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 2 Jul 2013 17:26:00 +0800 Subject: ASoC: imx_mc13783: add .owner to struct snd_soc_card Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/fsl/imx-mc13783.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 9df173c091a6..a3d60d4bea4c 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -90,6 +90,7 @@ static const struct snd_soc_dapm_route imx_mc13783_routes[] = { static struct snd_soc_card imx_mc13783 = { .name = "imx_mc13783", + .owner = THIS_MODULE, .dai_link = imx_mc13783_dai_mc13783, .num_links = ARRAY_SIZE(imx_mc13783_dai_mc13783), .dapm_widgets = imx_mc13783_widget, -- cgit v1.2.3-70-g09d2 From c364796a473db467b9201ea31a096bc0cf23547a Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Thu, 20 Jun 2013 15:20:20 +0200 Subject: ASoC: imx-pcm-dma: DT support This patch removes the NO_DT flag. The pdev pointer may have a proper of_node with the dmas property, so we can use it to request DMA channels. Signed-off-by: Markus Pargmann Tested-by: Shawn Guo Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-dma.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index fde4d2ea68c8..f323ce09f881 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -64,7 +64,6 @@ int imx_pcm_dma_init(struct platform_device *pdev) { return snd_dmaengine_pcm_register(&pdev->dev, &imx_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } EXPORT_SYMBOL_GPL(imx_pcm_dma_init); -- cgit v1.2.3-70-g09d2 From 9051cba110000985c1a50374fea16f1493955b6e Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Thu, 20 Jun 2013 15:20:21 +0200 Subject: ASoC: imx-pcm-fiq: Introduce pcm-fiq-params Cleaner parameter passing for imx-pcm-fiq. Create a seperated fiq-params struct to pass all arguments. Signed-off-by: Markus Pargmann Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-fiq.c | 18 ++++++++++-------- sound/soc/fsl/imx-pcm.h | 15 +++++++++++++-- sound/soc/fsl/imx-ssi.c | 7 ++++++- sound/soc/fsl/imx-ssi.h | 1 + 4 files changed, 30 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 310d90290320..3b2ba994beee 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -22,6 +22,7 @@ #include #include +#include #include #include #include @@ -32,6 +33,7 @@ #include #include "imx-ssi.h" +#include "imx-pcm.h" struct imx_pcm_runtime_data { unsigned int period; @@ -366,9 +368,9 @@ static struct snd_soc_platform_driver imx_soc_platform_fiq = { .pcm_free = imx_pcm_fiq_free, }; -int imx_pcm_fiq_init(struct platform_device *pdev) +int imx_pcm_fiq_init(struct platform_device *pdev, + struct imx_pcm_fiq_params *params) { - struct imx_ssi *ssi = platform_get_drvdata(pdev); int ret; ret = claim_fiq(&fh); @@ -377,15 +379,15 @@ int imx_pcm_fiq_init(struct platform_device *pdev) return ret; } - mxc_set_irq_fiq(ssi->irq, 1); - ssi_irq = ssi->irq; + mxc_set_irq_fiq(params->irq, 1); + ssi_irq = params->irq; - imx_pcm_fiq = ssi->irq; + imx_pcm_fiq = params->irq; - imx_ssi_fiq_base = (unsigned long)ssi->base; + imx_ssi_fiq_base = (unsigned long)params->base; - ssi->dma_params_tx.maxburst = 4; - ssi->dma_params_rx.maxburst = 6; + params->dma_params_tx->maxburst = 4; + params->dma_params_rx->maxburst = 6; ret = snd_soc_register_platform(&pdev->dev, &imx_soc_platform_fiq); if (ret) diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 67f656c7c320..fd56cad43cd6 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -32,6 +32,15 @@ imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, dma_data->peripheral_type = IMX_DMATYPE_SSI; } +struct imx_pcm_fiq_params { + int irq; + void __iomem *base; + + /* Pointer to original ssi driver to setup tx rx sizes */ + struct snd_dmaengine_dai_dma_data *dma_params_rx; + struct snd_dmaengine_dai_dma_data *dma_params_tx; +}; + #ifdef CONFIG_SND_SOC_IMX_PCM_DMA int imx_pcm_dma_init(struct platform_device *pdev); void imx_pcm_dma_exit(struct platform_device *pdev); @@ -47,10 +56,12 @@ static inline void imx_pcm_dma_exit(struct platform_device *pdev) #endif #ifdef CONFIG_SND_SOC_IMX_PCM_FIQ -int imx_pcm_fiq_init(struct platform_device *pdev); +int imx_pcm_fiq_init(struct platform_device *pdev, + struct imx_pcm_fiq_params *params); void imx_pcm_fiq_exit(struct platform_device *pdev); #else -static inline int imx_pcm_fiq_init(struct platform_device *pdev) +static inline int imx_pcm_fiq_init(struct platform_device *pdev, + struct imx_pcm_fiq_params *params) { return -ENODEV; } diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 51be3772cba9..f029e27366de 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -595,7 +595,12 @@ static int imx_ssi_probe(struct platform_device *pdev) goto failed_register; } - ret = imx_pcm_fiq_init(pdev); + ssi->fiq_params.irq = ssi->irq; + ssi->fiq_params.base = ssi->base; + ssi->fiq_params.dma_params_rx = &ssi->dma_params_rx; + ssi->fiq_params.dma_params_tx = &ssi->dma_params_tx; + + ret = imx_pcm_fiq_init(pdev, &ssi->fiq_params); if (ret) goto failed_pcm_fiq; diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index d5003cefca8d..fb1616ba8c59 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -209,6 +209,7 @@ struct imx_ssi { struct snd_dmaengine_dai_dma_data dma_params_tx; struct imx_dma_data filter_data_tx; struct imx_dma_data filter_data_rx; + struct imx_pcm_fiq_params fiq_params; int enabled; }; -- cgit v1.2.3-70-g09d2 From b2c119b0bba808608c48a8f7c9d727956d56561a Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 10 Jul 2013 18:43:54 +0800 Subject: ASoC: fsl: Disable SSI in trigger() if RE/TE are both cleared The code enabled SSIEN when triggered by SNDRV_PCM_TRIGGER_START, so move the disable code to SNDRV_PCM_TRIGGER_STOP for symmetric. This also allows us to use the SSI driver more flexible so that it can support some use cases like "aplay S16_LE.wav S24_LE.wav" which would call the driver in sequence like: startup()->hw_params(S16_LE)->trigger(START)->tirgger(STOP)-> hw_params(S24_LE)->trigger(START)->tirgger(STOP)->shutdown() If we disable SSIEN in shutdown(), the second hw_params() would bypass the sample bits setting while using symmetric_rate. Signed-off-by: Nicolin Chen Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837df07f..b6ab341a875c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -510,6 +510,9 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); else write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); + + if ((read_ssi(&ssi->scr) & (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); break; default: @@ -534,15 +537,6 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, ssi_private->first_stream = ssi_private->second_stream; ssi_private->second_stream = NULL; - - /* - * If this is the last active substream, disable the SSI. - */ - if (!ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - - write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); - } } static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) -- cgit v1.2.3-70-g09d2 From b641edfbf253a67a87698df275e3a35a7deac5d4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Jul 2013 16:31:58 +0100 Subject: ASoC: pcm3008: Remove noisy version print The version number has never been updated and the printk isn't based on any interaction with the device. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f2a6282b41f4..32e5a591f921 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,8 +28,6 @@ #include "pcm3008.h" -#define PCM3008_VERSION "0.2" - #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -64,8 +62,6 @@ static int pcm3008_soc_probe(struct snd_soc_codec *codec) struct pcm3008_setup_data *setup = codec->dev->platform_data; int ret = 0; - printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); - /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON * Low High De-emphasis OFF -- cgit v1.2.3-70-g09d2 From d7f184958e5292280c1048df951932bbfcd95f60 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Jul 2013 16:48:24 +0100 Subject: ASoC: pcm3008: Move gpio allocation to probe This is better from a device model point of view since we don't try to do things like instantiate the card until the required resources appear. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 87 ++++++++++++++++++++-------------------------- 1 file changed, 38 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 32e5a591f921..8ab1c03d26bd 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -57,10 +57,40 @@ static void pcm3008_gpio_free(struct pcm3008_setup_data *setup) gpio_free(setup->pdda_pin); } -static int pcm3008_soc_probe(struct snd_soc_codec *codec) +#ifdef CONFIG_PM +static int pcm3008_soc_suspend(struct snd_soc_codec *codec) +{ + struct pcm3008_setup_data *setup = codec->dev->platform_data; + + gpio_set_value(setup->pdad_pin, 0); + gpio_set_value(setup->pdda_pin, 0); + + return 0; +} + +static int pcm3008_soc_resume(struct snd_soc_codec *codec) { struct pcm3008_setup_data *setup = codec->dev->platform_data; - int ret = 0; + + gpio_set_value(setup->pdad_pin, 1); + gpio_set_value(setup->pdda_pin, 1); + + return 0; +} +#else +#define pcm3008_soc_suspend NULL +#define pcm3008_soc_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { + .suspend = pcm3008_soc_suspend, + .resume = pcm3008_soc_resume, +}; + +static int pcm3008_codec_probe(struct platform_device *pdev) +{ + struct pcm3008_setup_data *setup = pdev->dev.platform_data; + int ret; /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON @@ -97,63 +127,22 @@ static int pcm3008_soc_probe(struct snd_soc_codec *codec) if (ret != 0) goto gpio_err; - return ret; + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_pcm3008, &pcm3008_dai, 1); gpio_err: pcm3008_gpio_free(setup); - return ret; } -static int pcm3008_soc_remove(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - pcm3008_gpio_free(setup); - return 0; -} - -#ifdef CONFIG_PM -static int pcm3008_soc_suspend(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - gpio_set_value(setup->pdad_pin, 0); - gpio_set_value(setup->pdda_pin, 0); - - return 0; -} - -static int pcm3008_soc_resume(struct snd_soc_codec *codec) +static int pcm3008_codec_remove(struct platform_device *pdev) { - struct pcm3008_setup_data *setup = codec->dev->platform_data; + struct pcm3008_setup_data *setup = pdev->dev.platform_data; - gpio_set_value(setup->pdad_pin, 1); - gpio_set_value(setup->pdda_pin, 1); - - return 0; -} -#else -#define pcm3008_soc_suspend NULL -#define pcm3008_soc_resume NULL -#endif - -static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { - .probe = pcm3008_soc_probe, - .remove = pcm3008_soc_remove, - .suspend = pcm3008_soc_suspend, - .resume = pcm3008_soc_resume, -}; + snd_soc_unregister_codec(&pdev->dev); -static int pcm3008_codec_probe(struct platform_device *pdev) -{ - return snd_soc_register_codec(&pdev->dev, - &soc_codec_dev_pcm3008, &pcm3008_dai, 1); -} + pcm3008_gpio_free(setup); -static int pcm3008_codec_remove(struct platform_device *pdev) -{ - snd_soc_unregister_codec(&pdev->dev); return 0; } -- cgit v1.2.3-70-g09d2 From 33319a2fcc4482a97a8924e777bc0036d1d96696 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Jul 2013 16:49:16 +0100 Subject: ASoC: pcm3008: Check for platform data The driver will crash if none is provided. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 8ab1c03d26bd..4fa4ded30407 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -92,6 +92,9 @@ static int pcm3008_codec_probe(struct platform_device *pdev) struct pcm3008_setup_data *setup = pdev->dev.platform_data; int ret; + if (!setup) + return -EINVAL; + /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON * Low High De-emphasis OFF -- cgit v1.2.3-70-g09d2 From d57a79acc77160089e191b6c78e9d42bed517a62 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Jul 2013 16:53:55 +0100 Subject: ASoC: pcm3008: Convert to devm_gpio_request_one() Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 44 ++++++++++++-------------------------------- 1 file changed, 12 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 4fa4ded30407..b883f99d6f9f 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -49,14 +49,6 @@ static struct snd_soc_dai_driver pcm3008_dai = { }, }; -static void pcm3008_gpio_free(struct pcm3008_setup_data *setup) -{ - gpio_free(setup->dem0_pin); - gpio_free(setup->dem1_pin); - gpio_free(setup->pdad_pin); - gpio_free(setup->pdda_pin); -} - #ifdef CONFIG_PM static int pcm3008_soc_suspend(struct snd_soc_codec *codec) { @@ -103,49 +95,37 @@ static int pcm3008_codec_probe(struct platform_device *pdev) */ /* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */ - ret = gpio_request(setup->dem0_pin, "codec_dem0"); - if (ret == 0) - ret = gpio_direction_output(setup->dem0_pin, 1); + ret = devm_gpio_request_one(&pdev->dev, setup->dem0_pin, + GPIOF_OUT_INIT_HIGH, "codec_dem0"); if (ret != 0) - goto gpio_err; + return ret; /* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */ - ret = gpio_request(setup->dem1_pin, "codec_dem1"); - if (ret == 0) - ret = gpio_direction_output(setup->dem1_pin, 0); + ret = devm_gpio_request_one(&pdev->dev, setup->dem1_pin, + GPIOF_OUT_INIT_LOW, "codec_dem1"); if (ret != 0) - goto gpio_err; + return ret; /* Configure PDAD GPIO. */ - ret = gpio_request(setup->pdad_pin, "codec_pdad"); - if (ret == 0) - ret = gpio_direction_output(setup->pdad_pin, 1); + ret = devm_gpio_request_one(&pdev->dev, setup->pdad_pin, + GPIOF_OUT_INIT_HIGH, "codec_pdad"); if (ret != 0) - goto gpio_err; + return ret; /* Configure PDDA GPIO. */ - ret = gpio_request(setup->pdda_pin, "codec_pdda"); - if (ret == 0) - ret = gpio_direction_output(setup->pdda_pin, 1); + ret = devm_gpio_request_one(&pdev->dev, setup->pdda_pin, + GPIOF_OUT_INIT_HIGH, "codec_pdda"); if (ret != 0) - goto gpio_err; + return ret; return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm3008, &pcm3008_dai, 1); - -gpio_err: - pcm3008_gpio_free(setup); - return ret; } static int pcm3008_codec_remove(struct platform_device *pdev) { - struct pcm3008_setup_data *setup = pdev->dev.platform_data; - snd_soc_unregister_codec(&pdev->dev); - pcm3008_gpio_free(setup); - return 0; } -- cgit v1.2.3-70-g09d2 From 0d6178662cf3dde6829ace63408f25cab42e21c3 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 2 Jul 2013 17:26:21 +0800 Subject: ASoC: brownstone: add .owner to struct snd_soc_card Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/pxa/brownstone.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 4ad76099dd43..5b7d969f89a9 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -129,6 +129,7 @@ static struct snd_soc_dai_link brownstone_wm8994_dai[] = { /* audio machine driver */ static struct snd_soc_card brownstone = { .name = "brownstone", + .owner = THIS_MODULE, .dai_link = brownstone_wm8994_dai, .num_links = ARRAY_SIZE(brownstone_wm8994_dai), -- cgit v1.2.3-70-g09d2 From 3986b9829f638c8a7864e81cdb066aa3d79f05a5 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 2 Jul 2013 17:26:40 +0800 Subject: ASoC: ttc_dkb: add .owner to struct snd_soc_card Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/pxa/ttc-dkb.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index f4ea4f6663a2..13c9ee0cb83b 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -122,6 +122,7 @@ static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = { /* ttc/td audio machine driver */ static struct snd_soc_card ttc_dkb_card = { .name = "ttc-dkb-hifi", + .owner = THIS_MODULE, .dai_link = ttc_pm860x_hifi_dai, .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai), -- cgit v1.2.3-70-g09d2 From e7c8c589bb640a86685ee040e683b1ec93339f13 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 2 Jul 2013 17:25:37 +0800 Subject: ASoC: mop500: add .owner to struct snd_soc_card Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/ux500/mop500.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 8f5cd00a6e46..178d1bad6259 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -52,6 +52,7 @@ static struct snd_soc_dai_link mop500_dai_links[] = { static struct snd_soc_card mop500_card = { .name = "MOP500-card", + .owner = THIS_MODULE, .probe = NULL, .dai_link = mop500_dai_links, .num_links = ARRAY_SIZE(mop500_dai_links), -- cgit v1.2.3-70-g09d2 From 57e265c8d71fb94c130bfb31f589cc9e97fb3928 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Jul 2013 20:46:34 +0100 Subject: ASoC: wm8994: Move runtime PM init to platform device init As well as being better style this allows the device to idle when there is no audio card instantaited which is probably what we want. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1d4b1ec66e36..02c320f71cdf 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4014,9 +4014,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->micdet_irq = control->pdata.micdet_irq; - pm_runtime_enable(codec->dev); - pm_runtime_idle(codec->dev); - /* By default use idle_bias_off, will override for WM8994 */ codec->dapm.idle_bias_off = 1; @@ -4389,8 +4386,6 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); - pm_runtime_disable(codec->dev); - for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i, &wm8994->fll_locked[i]); @@ -4449,6 +4444,9 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8994, wm8994_dai, ARRAY_SIZE(wm8994_dai)); } @@ -4456,6 +4454,8 @@ static int wm8994_probe(struct platform_device *pdev) static int wm8994_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + return 0; } -- cgit v1.2.3-70-g09d2 From cb23e852aabb50f5083fb734c2067220d087d26e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 4 Jul 2013 20:01:01 -0300 Subject: ASoC: sglt5000: Provide the reg_stride field sgtl5000 has 16-bit registers, and only even numbers are valid for its registers addresses. Let regmap knows about this feature by specifying the 'reg_stride' field, so that it can access only the valid registers. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559dc92c..7c99f3ccb1c6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1470,6 +1470,7 @@ static struct snd_soc_codec_driver sgtl5000_driver = { static const struct regmap_config sgtl5000_regmap = { .reg_bits = 16, .val_bits = 16, + .reg_stride = 2, .max_register = SGTL5000_MAX_REG_OFFSET, .volatile_reg = sgtl5000_volatile, -- cgit v1.2.3-70-g09d2 From c50c2f7af1fe503b1c3b8cb0fa0e3c9b4bba9e92 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 15 Jul 2013 16:39:09 +0100 Subject: ASoC: kirkwood: Remove unused headers Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-openrd.c | 2 -- sound/soc/kirkwood/kirkwood-t5325.c | 2 -- 2 files changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index b979c7154715..addbebc2b3fa 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -16,9 +16,7 @@ #include #include #include -#include #include -#include #include "../codecs/cs42l51.h" static int openrd_client_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 1d0ed6f8add7..4f4cb56f765a 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -15,9 +15,7 @@ #include #include #include -#include #include -#include #include "../codecs/alc5623.h" static int t5325_hw_params(struct snd_pcm_substream *substream, -- cgit v1.2.3-70-g09d2 From 30d3924852c7b07df4e015610aca1cafa5c15cab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 15 Jul 2013 16:40:18 +0100 Subject: ASoC: kirkwood: Enable build on non-Kirkwood platforms Improve build coverage by enabling build on other platforms if COMPILE_TEST is enabled. Signed-off-by: Mark Brown --- sound/soc/kirkwood/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index c62d715235e2..59085ad6c41a 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood chip" - depends on ARCH_KIRKWOOD + depends on ARCH_KIRKWOOD || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the @@ -11,7 +11,7 @@ config SND_KIRKWOOD_SOC_I2S config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" - depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) + depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) depends on I2C select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 @@ -21,7 +21,7 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C + depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C select SND_KIRKWOOD_SOC_I2S select SND_SOC_ALC5623 help -- cgit v1.2.3-70-g09d2 From 4734dc96ea50109ae08603af10804f989ff35437 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 15 Jul 2013 16:41:14 +0100 Subject: ASoC: kirkwood-i2s: Use devm_clk_get() for extclk Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3263c5..44412eaf6e81 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -498,10 +498,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (err < 0) return err; - priv->extclk = clk_get(&pdev->dev, "extclk"); + priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { - clk_put(priv->extclk); priv->extclk = ERR_PTR(-EINVAL); } else { dev_info(&pdev->dev, "found external clock\n"); @@ -529,10 +528,8 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return 0; dev_err(&pdev->dev, "snd_soc_register_component failed\n"); - if (!IS_ERR(priv->extclk)) { + if (!IS_ERR(priv->extclk)) clk_disable_unprepare(priv->extclk); - clk_put(priv->extclk); - } clk_disable_unprepare(priv->clk); return err; @@ -544,10 +541,8 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); - if (!IS_ERR(priv->extclk)) { + if (!IS_ERR(priv->extclk)) clk_disable_unprepare(priv->extclk); - clk_put(priv->extclk); - } clk_disable_unprepare(priv->clk); return 0; -- cgit v1.2.3-70-g09d2 From 60478295d6876619f8f47f6d1a5c25eaade69ee3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:55:57 +0200 Subject: ALSA: asihpi: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 185d54a5cb1a..dc632cdc3870 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -769,7 +769,10 @@ static void snd_card_asihpi_timer_function(unsigned long data) s->number); ds->drained_count++; if (ds->drained_count > 20) { + unsigned long flags; + snd_pcm_stream_lock_irqsave(s, flags); snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(s, flags); continue; } } else { -- cgit v1.2.3-70-g09d2 From cc7282b8d5abbd48c81d1465925d464d9e3eaa8f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:56:56 +0200 Subject: ALSA: atiixp: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Signed-off-by: Takashi Iwai --- sound/pci/atiixp.c | 2 ++ sound/pci/atiixp_modem.c | 2 ++ 2 files changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index fe4c61bdb8ba..f6dec3ea371f 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -689,7 +689,9 @@ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma) if (! dma->substream || ! dma->running) return; snd_printdd("atiixp: XRUN detected (DMA %d)\n", dma->ops->type); + snd_pcm_stream_lock(dma->substream); snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(dma->substream); } /* diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index cf29b9a1d65d..289563ecb6dd 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -638,7 +638,9 @@ static void snd_atiixp_xrun_dma(struct atiixp_modem *chip, if (! dma->substream || ! dma->running) return; snd_printdd("atiixp-modem: XRUN detected (DMA %d)\n", dma->ops->type); + snd_pcm_stream_lock(dma->substream); snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(dma->substream); } /* -- cgit v1.2.3-70-g09d2 From 5b9ab3f7324a1b94a5a5a76d44cf92dfeb3b5e80 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:57:55 +0200 Subject: ALSA: 6fire: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index c5b9cac37dc4..2aa4e13063a8 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -639,17 +639,25 @@ int usb6fire_pcm_init(struct sfire_chip *chip) void usb6fire_pcm_abort(struct sfire_chip *chip) { struct pcm_runtime *rt = chip->pcm; + unsigned long flags; int i; if (rt) { rt->panic = true; - if (rt->playback.instance) + if (rt->playback.instance) { + snd_pcm_stream_lock_irqsave(rt->playback.instance, flags); snd_pcm_stop(rt->playback.instance, SNDRV_PCM_STATE_XRUN); - if (rt->capture.instance) + snd_pcm_stream_unlock_irqrestore(rt->playback.instance, flags); + } + + if (rt->capture.instance) { + snd_pcm_stream_lock_irqsave(rt->capture.instance, flags); snd_pcm_stop(rt->capture.instance, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(rt->capture.instance, flags); + } for (i = 0; i < PCM_N_URBS; i++) { usb_poison_urb(&rt->in_urbs[i].instance); -- cgit v1.2.3-70-g09d2 From cb6f66a2d278e57a6c9d8fb59bd9ebd8ab3965c2 Mon Sep 17 00:00:00 2001 From: Chih-Chung Chang Date: Mon, 15 Jul 2013 09:38:46 -0700 Subject: ASoC: max98088 - fix element type of the register cache. The registers of max98088 are 8 bits, not 16 bits. This bug causes the contents of registers to be overwritten with bad values when the codec is suspended and then resumed. Signed-off-by: Chih-Chung Chang Signed-off-by: Dylan Reid Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/max98088.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3eeada57e87d..566a367c94fa 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) static void max98088_sync_cache(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; + u8 *reg_cache = codec->reg_cache; int i; if (!codec->cache_sync) -- cgit v1.2.3-70-g09d2 From 9538aa46c2427d6782aa10036c4da4c541605e0e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:58:25 +0200 Subject: ALSA: ua101: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/misc/ua101.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 8b5d2c564e04..509315937f25 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -613,14 +613,24 @@ static int start_usb_playback(struct ua101 *ua) static void abort_alsa_capture(struct ua101 *ua) { - if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + unsigned long flags; + + if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) { + snd_pcm_stream_lock_irqsave(ua->capture.substream, flags); snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(ua->capture.substream, flags); + } } static void abort_alsa_playback(struct ua101 *ua) { - if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + unsigned long flags; + + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) { + snd_pcm_stream_lock_irqsave(ua->playback.substream, flags); snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(ua->playback.substream, flags); + } } static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, -- cgit v1.2.3-70-g09d2 From 5be1efb4c2ed79c3d7c0cbcbecae768377666e84 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:58:47 +0200 Subject: ALSA: usx2y: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usbusx2yaudio.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 4967fe9c938d..63fb5219f0f8 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -273,7 +273,11 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) struct snd_usX2Y_substream *subs = usX2Y->subs[s]; if (subs) { if (atomic_read(&subs->state) >= state_PRERUNNING) { + unsigned long flags; + + snd_pcm_stream_lock_irqsave(subs->pcm_substream, flags); snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(subs->pcm_substream, flags); } for (u = 0; u < NRURBS; u++) { struct urb *urb = subs->urb[u]; -- cgit v1.2.3-70-g09d2 From 46f6c1aaf790be9ea3c8ddfc8f235a5f677d08e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:59:33 +0200 Subject: ALSA: pxa2xx: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-pcm-lib.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 76e0d5695075..823359ed95e1 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -166,7 +166,9 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) } else { printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", rtd->params->name, dma_ch, dcsr); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); } } EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); -- cgit v1.2.3-70-g09d2 From 571185717f8d7f2a088a7ac38d94a9ad5fd9da5c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 18:00:01 +0200 Subject: ASoC: atmel: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/atmel/atmel-pcm-dma.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 1d38fd0bc4e2..d12826526798 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -81,7 +81,9 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, /* stop RX and capture: will be enabled again at restart */ ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); /* now drain RHR and read status to remove xrun condition */ ssc_readx(prtd->ssc->regs, SSC_RHR); -- cgit v1.2.3-70-g09d2 From 61be2b9a18ec70f3cbe3deef7a5f77869c71b5ae Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 18:00:25 +0200 Subject: ASoC: s6000: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s6000/s6000-pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 1358c7de2521..d0740a762963 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -128,7 +128,9 @@ static irqreturn_t s6000_pcm_irq(int irq, void *data) substream->runtime && snd_pcm_running(substream)) { dev_dbg(pcm->dev, "xrun\n"); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); ret = IRQ_HANDLED; } -- cgit v1.2.3-70-g09d2 From d52392b1a80458c0510810789c7db4a39b88022a Mon Sep 17 00:00:00 2001 From: Aaron Plattner Date: Fri, 12 Jul 2013 11:01:37 -0700 Subject: ALSA: hda - Add new GPU codec ID to snd-hda Vendor ID 0x10de0060 is used by a yet-to-be-named GPU chip. Reviewed-by: Andy Ritger Signed-off-by: Aaron Plattner Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 540bdef2f904..030ca8652a1c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2622,6 +2622,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -2674,6 +2675,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0042"); MODULE_ALIAS("snd-hda-codec-id:10de0043"); MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0051"); +MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); -- cgit v1.2.3-70-g09d2 From 6ad74047f47e1c48223237a0032c5e000f01193f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Jul 2013 10:46:16 +0100 Subject: ASoC: kirkwood-i2s: Remove empty remove() Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 44412eaf6e81..becc082e7ee3 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -398,11 +398,6 @@ static int kirkwood_i2s_probe(struct snd_soc_dai *dai) } -static int kirkwood_i2s_remove(struct snd_soc_dai *dai) -{ - return 0; -} - static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .startup = kirkwood_i2s_startup, .trigger = kirkwood_i2s_trigger, @@ -413,7 +408,6 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { static struct snd_soc_dai_driver kirkwood_i2s_dai = { .probe = kirkwood_i2s_probe, - .remove = kirkwood_i2s_remove, .playback = { .channels_min = 1, .channels_max = 2, @@ -431,7 +425,6 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .probe = kirkwood_i2s_probe, - .remove = kirkwood_i2s_remove, .playback = { .channels_min = 1, .channels_max = 2, -- cgit v1.2.3-70-g09d2 From 9e12cbd93232c20544d16aa33c587786a6cb726d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Jul 2013 10:47:10 +0100 Subject: ASoC: kirkwood-i2s: Inline KIRKWOOD_I2S_RATES The addition of extclk support makes this misleading as it's only the rates used when there is no extclk so put it in the specific DAI it applies to. Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index becc082e7ee3..e6027fd5b29c 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -26,9 +26,6 @@ #define DRV_NAME "kirkwood-i2s" -#define KIRKWOOD_I2S_RATES \ - (SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) #define KIRKWOOD_I2S_FORMATS \ (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ @@ -411,13 +408,15 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = KIRKWOOD_I2S_RATES, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, .formats = KIRKWOOD_I2S_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, - .rates = KIRKWOOD_I2S_RATES, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, -- cgit v1.2.3-70-g09d2 From 8a537f85e9db8a43b323b0ffcf358c51448491de Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Tue, 16 Jul 2013 08:47:47 +0200 Subject: ASoC: kirkwood-i2s: fix a compilation warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In the function kirkwood_set_rate, when the rate cannot be satisfied by the internal nor by an external clock, the clock source in undefined: warning: ‘clks_ctrl’ may be used uninitialized in this function The ALSA subsystem should never gives such a rate because: - the rates with the internal clock are limited to 44.1, 48 and 96 kHz as specified by the kirkwood_i2s_dai structure, - the other rates are proposed in the structure kirkwood_i2s_dai_extclk only when the external clock is present. In case of programming error (bad rate for internal clock and no external clock), the function will simply cause a backtrace. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index e6027fd5b29c..ba7203995369 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -102,14 +102,16 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, uint32_t clks_ctrl; if (rate == 44100 || rate == 48000 || rate == 96000) { - /* use internal dco for supported rates */ + /* use internal dco for the supported rates + * defined in kirkwood_i2s_dai */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", __func__, rate); kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; - } else if (!IS_ERR(priv->extclk)) { - /* use optional external clk for other rates */ + } else { + /* use the external clock for the other rates + * defined in kirkwood_i2s_dai_extclk */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate); clk_set_rate(priv->extclk, 256 * rate); -- cgit v1.2.3-70-g09d2 From f2c4fa655f7139a181a6d6db99a49cab96ed0337 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Jul 2013 13:36:05 +0100 Subject: ASoC: tlv320aic3x: Add compatible strings for specific devices The driver supports a range of devices but currently doesn't allow those device names to be used for enumeration on DT. Add the currently listed I2C IDs as compatible strings. Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 8 +++++++- sound/soc/codecs/tlv320aic3x.c | 2 ++ 2 files changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index f47c3f589fd0..26f65f92f42d 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -3,7 +3,13 @@ Texas Instruments - tlv320aic3x Codec module The tlv320aic3x serial control bus communicates through I2C protocols Required properties: -- compatible - "string" - "ti,tlv320aic3x" + +- compatible - "string" - One of: + "ti,tlv320aic3x" - Generic TLV320AIC3x device + "ti,tlv320aic33" - TLV320AIC33 + "ti,tlv320aic3007" - TLV320AIC3007 + + - reg - - I2C slave address diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e5b926883131..c9bb760f405f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1582,6 +1582,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) #if defined(CONFIG_OF) static const struct of_device_id tlv320aic3x_of_match[] = { { .compatible = "ti,tlv320aic3x", }, + { .compatible = "ti,tlv320aic33" }, + { .compatible = "ti,tlv320aic3007" }, {}, }; MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); -- cgit v1.2.3-70-g09d2 From cbaa56896146cbb5ab54bd65f98d020af282e6c6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Jul 2013 13:39:52 +0100 Subject: ASoC: tlv320aic3x: List tlv320aic3106 as a supported device Currently there is no specific handling for it but the tlv320aic3106 is supported using this driver. Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 1 + sound/soc/codecs/tlv320aic3x.c | 2 ++ 2 files changed, 3 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 26f65f92f42d..705a6b156c6c 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -8,6 +8,7 @@ Required properties: "ti,tlv320aic3x" - Generic TLV320AIC3x device "ti,tlv320aic33" - TLV320AIC33 "ti,tlv320aic3007" - TLV320AIC3007 + "ti,tlv320aic3106" - TLV320AIC3106 - reg - - I2C slave address diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index c9bb760f405f..cad4fb17a933 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1492,6 +1492,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic3x", AIC3X_MODEL_3X }, { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, + { "tlv320aic3106", AIC3X_MODEL_3X }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1584,6 +1585,7 @@ static const struct of_device_id tlv320aic3x_of_match[] = { { .compatible = "ti,tlv320aic3x", }, { .compatible = "ti,tlv320aic33" }, { .compatible = "ti,tlv320aic3007" }, + { .compatible = "ti,tlv320aic3106" }, {}, }; MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); -- cgit v1.2.3-70-g09d2 From a1df5c2b310ff88cad66de6d55c06d4ad3b9684b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 15 Jul 2013 17:03:18 +0100 Subject: ASoC: imx: Enable COMPILE_TEST builds Since DT based boards don't have any dependency on arch/arm enable them if COMPILE_TEST is enabled. Signed-off-by: Mark Brown Acked-by: Shawn Guo --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa438546c912..87e28bcf7a83 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -98,7 +98,7 @@ endif # SND_POWERPC_SOC menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC + depends on ARCH_MXC || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the i.MX CPUs. -- cgit v1.2.3-70-g09d2 From 7497185f8c5a3e0dc92765bfc723900b492cd8a4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 15 Jul 2013 18:13:35 +0100 Subject: ASoC: samsung-spdif: Convert to devm_clk_get() Signed-off-by: Mark Brown --- sound/soc/samsung/spdif.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 2e5ebb2f1982..5ea70ab0ecb5 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -395,7 +395,7 @@ static int spdif_probe(struct platform_device *pdev) spin_lock_init(&spdif->lock); - spdif->pclk = clk_get(&pdev->dev, "spdif"); + spdif->pclk = devm_clk_get(&pdev->dev, "spdif"); if (IS_ERR(spdif->pclk)) { dev_err(&pdev->dev, "failed to get peri-clock\n"); ret = -ENOENT; @@ -403,7 +403,7 @@ static int spdif_probe(struct platform_device *pdev) } clk_prepare_enable(spdif->pclk); - spdif->sclk = clk_get(&pdev->dev, "sclk_spdif"); + spdif->sclk = devm_clk_get(&pdev->dev, "sclk_spdif"); if (IS_ERR(spdif->sclk)) { dev_err(&pdev->dev, "failed to get internal source clock\n"); ret = -ENOENT; @@ -457,10 +457,8 @@ err3: release_mem_region(mem_res->start, resource_size(mem_res)); err2: clk_disable_unprepare(spdif->sclk); - clk_put(spdif->sclk); err1: clk_disable_unprepare(spdif->pclk); - clk_put(spdif->pclk); err0: return ret; } @@ -480,9 +478,7 @@ static int spdif_remove(struct platform_device *pdev) release_mem_region(mem_res->start, resource_size(mem_res)); clk_disable_unprepare(spdif->sclk); - clk_put(spdif->sclk); clk_disable_unprepare(spdif->pclk); - clk_put(spdif->pclk); return 0; } -- cgit v1.2.3-70-g09d2 From f1269ae41f2e2846031cb361b59ebc36d5f9528a Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 16 Jul 2013 20:05:07 +0800 Subject: ASoC: imx-sgtl5000: fix error return code in imx_sgtl5000_probe() Fix to return a negative error code from the error handling case instead of 0, as done elsewhere in this function. Signed-off-by: Wei Yongjun Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/imx-sgtl5000.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 3f726e4f88db..389cbfa6dca7 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -129,8 +129,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); goto fail; + } data->clk_frequency = clk_get_rate(data->codec_clk); -- cgit v1.2.3-70-g09d2 From 46a5905e1cd4a9d9d238ec7beece49ce49e2ad85 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Tue, 16 Jul 2013 09:17:27 +0800 Subject: ASoC: sgtl5000: defer the probe if clock is not found It's not always the case that clock is already available when sgtl5000 get probed at the first time, e.g. the clock is provided by CPU DAI which may be probed after sgtl5000. So let's defer the probe when devm_clk_get() call fails and give it chance to try later. It fixes the regression on imx28 since commit 9e13f34 (ASoC: sgtl5000: Let the codec acquire its clock). [ 1.927637] sgtl5000 0-000a: Failed to get mclock: -2 [ 1.934280] sgtl5000: probe of 0-000a failed with error -2 [ 1.945906] mxs-sgtl5000 sound.13: ASoC: CODEC (null) not registered [ 1.953787] mxs-sgtl5000 sound.13: snd_soc_register_card failed (-517) [ 1.960865] platform sound.13: Driver mxs-sgtl5000 requests probe deferral Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d659d3adcfb3..6c8a9e7bee25 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1527,6 +1527,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); dev_err(&client->dev, "Failed to get mclock: %d\n", ret); + /* Defer the probe to see if the clk will be provided later */ + if (ret == -ENOENT) + return -EPROBE_DEFER; return ret; } -- cgit v1.2.3-70-g09d2 From 256ca9c3ad5013ff8a8f165e5a82fab437628c8e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Jul 2013 12:17:49 +0200 Subject: ALSA: seq-oss: Initialize MIDI clients asynchronously We've got bug reports that the module loading stuck on Debian system with 3.10 kernel. The debugging session revealed that the initial registration of OSS sequencer clients stuck at module loading time, which involves again with request_module() at the init phase. This is triggered only by special --install stuff Debian is using, but it's still not good to have such loops. As a workaround, call the registration part asynchronously. This is a better approach irrespective of the hang fix, in anyway. Reported-and-tested-by: Philipp Matthias Hahn Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_init.c | 16 +++++++++++++--- sound/core/seq/oss/seq_oss_midi.c | 2 +- 2 files changed, 14 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index e3cb46fef2c7..b3f39b5ed742 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -31,6 +31,7 @@ #include #include #include +#include /* * common variables @@ -60,6 +61,14 @@ static void free_devinfo(void *private); #define call_ctl(type,rec) snd_seq_kernel_client_ctl(system_client, type, rec) +/* call snd_seq_oss_midi_lookup_ports() asynchronously */ +static void async_call_lookup_ports(struct work_struct *work) +{ + snd_seq_oss_midi_lookup_ports(system_client); +} + +static DECLARE_WORK(async_lookup_work, async_call_lookup_ports); + /* * create sequencer client for OSS sequencer */ @@ -85,9 +94,6 @@ snd_seq_oss_create_client(void) system_client = rc; debug_printk(("new client = %d\n", rc)); - /* look up midi devices */ - snd_seq_oss_midi_lookup_ports(system_client); - /* create annoucement receiver port */ memset(port, 0, sizeof(*port)); strcpy(port->name, "Receiver"); @@ -115,6 +121,9 @@ snd_seq_oss_create_client(void) } rc = 0; + /* look up midi devices */ + schedule_work(&async_lookup_work); + __error: kfree(port); return rc; @@ -160,6 +169,7 @@ receive_announce(struct snd_seq_event *ev, int direct, void *private, int atomic int snd_seq_oss_delete_client(void) { + cancel_work_sync(&async_lookup_work); if (system_client >= 0) snd_seq_delete_kernel_client(system_client); diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 677dc84590c7..862d84893ee8 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -72,7 +72,7 @@ static int send_midi_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, * look up the existing ports * this looks a very exhausting job. */ -int __init +int snd_seq_oss_midi_lookup_ports(int client) { struct snd_seq_client_info *clinfo; -- cgit v1.2.3-70-g09d2 From b0e0a4d6faaa0415ecc1fa6b9a08dd17df05edad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 15 Jul 2013 16:56:32 +0100 Subject: ASoC: omap: Enable COMPILE_TEST build for DT platforms The DT platforms don't have any source dependency on any OMAP stuff so allow them to be built when COMPILE_TEST is enabled. Signed-off-by: Mark Brown Acked-by: Peter Ujfalusi --- sound/soc/omap/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 9f5d55e6b17a..44b5acf01f3f 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,6 +1,6 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on ARCH_OMAP && DMA_OMAP + depends on (ARCH_OMAP && DMA_OMAP) || (ARCH_ARM && COMPILE_TEST) select SND_SOC_DMAENGINE_PCM config SND_OMAP_SOC_DMIC @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && MACH_NOKIA_RX51 + depends on SND_OMAP_SOC && ARCH_ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 @@ -87,7 +87,7 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 -- cgit v1.2.3-70-g09d2 From f9b4243fc2e0be109b957a0a5a25968facf7565d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Jul 2013 17:00:27 +0100 Subject: ASoC: tegra: Remove unneeded mach-type.h incldues Signed-off-by: Mark Brown Acked-by: Stephen Warren Tested-by: Stephen Warren --- sound/soc/tegra/tegra_alc5632.c | 2 -- sound/soc/tegra/tegra_wm8753.c | 2 -- sound/soc/tegra/trimslice.c | 2 -- 3 files changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 48d05d9e1002..c61ea3a1030f 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -13,8 +13,6 @@ * published by the Free Software Foundation. */ -#include - #include #include #include diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index f87fc53e9b8c..8e774d1a243c 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -28,8 +28,6 @@ * */ -#include - #include #include #include diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 05c68aab5cf0..734bfcd21148 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -24,8 +24,6 @@ * */ -#include - #include #include #include -- cgit v1.2.3-70-g09d2 From 2fa1b9008c73525bbd7de93bf36e406b8a754bd1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Jul 2013 18:49:09 +0100 Subject: ASoC: tegra: Add GPIOLIB dependencies For build coverage. Signed-off-by: Mark Brown Acked-by: Stephen Warren --- sound/soc/tegra/Kconfig | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 995b120c2cd0..b0c8ecf8101b 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -61,7 +61,7 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" - depends on SND_SOC_TEGRA && I2C + depends on SND_SOC_TEGRA && I2C && GPIOLIB select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_RT5640 @@ -71,7 +71,7 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" - depends on SND_SOC_TEGRA && I2C + depends on SND_SOC_TEGRA && I2C && GPIOLIB select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8753 @@ -81,7 +81,7 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" - depends on SND_SOC_TEGRA && I2C + depends on SND_SOC_TEGRA && I2C && GPIOLIB select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 @@ -92,7 +92,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB select SND_SOC_TEGRA20_AC97 select SND_SOC_WM9712 help @@ -110,7 +110,7 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" - depends on SND_SOC_TEGRA && I2C + depends on SND_SOC_TEGRA && I2C && GPIOLIB select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help -- cgit v1.2.3-70-g09d2 From c6c0925ea32d37696da7d71631a4a0c999f2094f Mon Sep 17 00:00:00 2001 From: Rongjun Ying Date: Wed, 17 Jul 2013 14:12:16 +0800 Subject: ASoC: hdmi-codec: let the driver support HDMI sink Devices like mobilephones, computers are typically used as HDMI sources, but devices like TV, navigators will be HDMI sinks. for auto scenerios, In-Vehicle Infotainment(IVI) can be HDMI sink to display movies from mobilephones. Signed-off-by: Rongjun Ying Signed-off-by: Barry Song Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index 2bcae2b40c92..f0986b9f1939 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -37,6 +37,17 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, + }; static int hdmi_codec_probe(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From b0a4747a5d6498d37ebb6e4ce53dd5d89c51ab51 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 17 Jul 2013 02:00:38 -0300 Subject: ASoC: fsl: fsl_ssi: Use devm_ functions Using devm_ functions can make the code cleaner and smaller. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 38 ++++++++++---------------------------- 1 file changed, 10 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b6ab341a875c..d078b1ba08e8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -674,7 +674,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* The DAI name is the last part of the full name of the node. */ p = strrchr(np->full_name, '/') + 1; - ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p), + ssi_private = devm_kzalloc(&pdev->dev, sizeof(*ssi_private) + strlen(p), GFP_KERNEL); if (!ssi_private) { dev_err(&pdev->dev, "could not allocate DAI object\n"); @@ -692,26 +692,24 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = of_address_to_resource(np, 0, &res); if (ret) { dev_err(&pdev->dev, "could not determine device resources\n"); - goto error_kmalloc; + return ret; } ssi_private->ssi = of_iomap(np, 0); if (!ssi_private->ssi) { dev_err(&pdev->dev, "could not map device resources\n"); - ret = -ENOMEM; - goto error_kmalloc; + return -ENOMEM; } ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); if (ssi_private->irq == NO_IRQ) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); - ret = -ENXIO; - goto error_iomap; + return -ENXIO; } /* The 'name' should not have any slashes in it. */ - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, - ssi_private); + ret = devm_request_irq(&pdev->dev, ssi_private->irq, fsl_ssi_isr, 0, + ssi_private->name, ssi_private); if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); goto error_irqmap; @@ -733,11 +731,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) u32 dma_events[2]; ssi_private->ssi_on_imx = true; - ssi_private->clk = clk_get(&pdev->dev, NULL); + ssi_private->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi_private->clk)) { ret = PTR_ERR(ssi_private->clk); dev_err(&pdev->dev, "could not get clock: %d\n", ret); - goto error_irq; + goto error_irqmap; } clk_prepare_enable(ssi_private->clk); @@ -788,7 +786,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); - goto error_irq; + goto error_clk; } /* Register with ASoC */ @@ -851,23 +849,12 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx) clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); - } - -error_irq: - free_irq(ssi_private->irq, ssi_private); error_irqmap: irq_dispose_mapping(ssi_private->irq); -error_iomap: - iounmap(ssi_private->ssi); - -error_kmalloc: - kfree(ssi_private); - return ret; } @@ -880,15 +867,10 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->ssi_on_imx) { imx_pcm_dma_exit(pdev); clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); } snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - - free_irq(ssi_private->irq, ssi_private); irq_dispose_mapping(ssi_private->irq); - - kfree(ssi_private); dev_set_drvdata(&pdev->dev, NULL); return 0; -- cgit v1.2.3-70-g09d2 From ede32d3a237e102884cd5b223aba9afe3e6fb679 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 17 Jul 2013 02:00:39 -0300 Subject: ASoC: fsl: fsl_ssi: Check the return value from clk_prepare_enable() clk_prepare_enable() may fail, so let's check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d078b1ba08e8..c9974a4ac042 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -737,7 +737,12 @@ static int fsl_ssi_probe(struct platform_device *pdev) dev_err(&pdev->dev, "could not get clock: %d\n", ret); goto error_irqmap; } - clk_prepare_enable(ssi_private->clk); + ret = clk_prepare_enable(ssi_private->clk); + if (ret) { + dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", + ret); + goto error_irqmap; + } /* * We have burstsize be "fifo_depth - 2" to match the SSI -- cgit v1.2.3-70-g09d2 From 2086d078359f0fa512543404f772fc0615da385a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 17 Jul 2013 10:18:33 +0100 Subject: ASoC: tegra: Always use the generic dmaengine helper library The usage of the dmaengine helpers is unconditional, especially when doing compile testing. Reported-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index b0c8ecf8101b..66b7a068202e 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -2,7 +2,7 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" depends on ARCH_TEGRA && TEGRA20_APB_DMA select REGMAP_MMIO - select SND_SOC_GENERIC_DMAENGINE_PCM if TEGRA20_APB_DMA + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want support for SoC audio on Tegra. -- cgit v1.2.3-70-g09d2 From 22abf843af0686a58b2b6b33d02388d4bbbbcd25 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 15 Jul 2013 17:09:40 +0100 Subject: ASoC: tegra: Enable COMPILE_TEST builds Since there is no architecture dependency in the code allow it to be built on any platform when COMPILE_TEST is enabled. Signed-off-by: Mark Brown Acked-by: Stephen Warren --- sound/soc/tegra/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 66b7a068202e..8fc653ca3ab4 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" - depends on ARCH_TEGRA && TEGRA20_APB_DMA + depends on (ARCH_TEGRA && TEGRA20_APB_DMA) || COMPILE_TEST select REGMAP_MMIO select SND_SOC_GENERIC_DMAENGINE_PCM help -- cgit v1.2.3-70-g09d2 From d0c05ad7827df545760e7659471965ce4ccf655d Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 16 Jul 2013 11:00:44 -0600 Subject: ASoC: tegra: fix compile warning in AC'97 driver MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This fixes the following by deleting dead code: sound/soc/tegra/tegra20_ac97.c: In function ‘tegra20_ac97_platform_probe’: sound/soc/tegra/tegra20_ac97.c:435:1: warning: label ‘err_unregister_pcm’ defined but not used [-Wunused-label] Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index e58233f7df61..87b845f4cd1b 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -432,8 +432,6 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) return 0; -err_unregister_pcm: - tegra_pcm_platform_unregister(&pdev->dev); err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_asoc_utils_fini: -- cgit v1.2.3-70-g09d2 From 02502da4579ffcd2b96334297ba8e6daefe618c4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 10:46:29 +0100 Subject: ASoC: imx-mc13783: Depend on ARCH_ARM The driver uses the machine type macros so depends on ARM. Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 87e28bcf7a83..3a79d01e8d03 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -194,7 +194,7 @@ config SND_SOC_IMX_SGTL5000 config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" - depends on MFD_MC13783 + depends on MFD_MC13783 && ARCH_ARM select SND_SOC_IMX_SSI select SND_SOC_IMX_AUDMUX select SND_SOC_MC13783 -- cgit v1.2.3-70-g09d2 From e94a093c1c7a1b06fa574c99f69ed594e0c52ff2 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Thu, 18 Jul 2013 14:38:20 +0800 Subject: ASoC: wm8904: fix the typo error for LINER Mux fix the typo error, from "LINEL Mux" to "LINER Mux" Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4c9fb142cb2d..91dfbfeda6f8 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1012,7 +1012,7 @@ static const struct soc_enum liner_enum = SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text); static const struct snd_kcontrol_new liner_mux = - SOC_DAPM_ENUM("LINEL Mux", liner_enum); + SOC_DAPM_ENUM("LINER Mux", liner_enum); static const char *sidetone_text[] = { "None", "Left", "Right" -- cgit v1.2.3-70-g09d2 From d4e1a73acd4e894f8332f2093bceaef585cfab67 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 11:52:17 +0100 Subject: ASoC: pcm: Use the power efficient workqueue for delayed powerdown There is no need to use a normal per-CPU workqueue for delayed power downs as they're not timing or performance critical and waking up a core for them would defeat some of the point. Signed-off-by: Mark Brown Reviewed-by: Viresh Kumar --- sound/soc/soc-pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b6c640332a17..f4f68cb3cb00 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -411,8 +411,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* start delayed pop wq here for playback streams */ rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ -- cgit v1.2.3-70-g09d2 From d8a14e302ffeecc312186b8b9b0efc8963cec83b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 18 Jul 2013 15:07:48 -0300 Subject: ASoC: fsl: imx-wm8962: Fix error path If the 'failed to find codec platform device' error path is executed, it should jump to 'fail' label instead of returning an error immediately. 'fail' label will then free the ssi_np and codec_np previously acquired nodes. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-wm8962.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 52a36a90f4f4..1d70e278e915 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -217,7 +217,8 @@ static int imx_wm8962_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev || !codec_dev->driver) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + ret = -EINVAL; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- cgit v1.2.3-70-g09d2 From 1ea9a69d1a36a5b62bf281ba8bb304fcac656dad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Jul 2013 07:58:02 +0200 Subject: ALSA: hda - Fix EAPD GPIO control for Sigmatel codecs The EAPD GPIO is dynamically turned on/off for some machines with Sigmatel codecs, but this didn't work as expected, and it resulted in spontaneous lost of speaker outputs per HP plugging or power-saving. This patch fixes the bug by simply including spec->eapd_mask into spec->gpio_mask and spec->gpio_data bits. Reported-and-tested-by: Eric Shattow Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e2f83591161b..766e56754c64 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -417,9 +417,11 @@ static void stac_update_outputs(struct hda_codec *codec) val &= ~spec->eapd_mask; else val |= spec->eapd_mask; - if (spec->gpio_data != val) + if (spec->gpio_data != val) { + spec->gpio_data = val; stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, val); + } } } @@ -3612,20 +3614,18 @@ static int stac_parse_auto_config(struct hda_codec *codec) static int stac_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int gpio; int i; /* override some hints */ stac_store_hints(codec); /* set up GPIO */ - gpio = spec->gpio_data; /* turn on EAPD statically when spec->eapd_switch isn't set. * otherwise, unsol event will turn it on/off dynamically */ if (!spec->eapd_switch) - gpio |= spec->eapd_mask; - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio); + spec->gpio_data |= spec->eapd_mask; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); snd_hda_gen_init(codec); @@ -3915,6 +3915,7 @@ static void stac_setup_gpio(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; + spec->gpio_mask |= spec->eapd_mask; if (spec->gpio_led) { if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; -- cgit v1.2.3-70-g09d2 From f3e351eef3a7fd1e36a3e18d4f2f069b00deb23c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Jul 2013 08:02:25 +0200 Subject: ALSA: hda - Remove NO_PRESENCE bit override for Dell 1420n Laptop The quirk for Dell laptops with STAC9228 overrides the pin default config of NID 0x0f to the value with AC_DEFCFG_MISC_NO_PRESENCE bit on. I'm not quite sure why this was done so, but can guess that this was introduced for avoiding this to be muted by another headphone plug. Now, after transition to the generic parser, this workaround rather causes a problem (notably as unexpected speaker mutes) because the pin is seen as if it's always plugged in. Since the generic parser can handle multiple headphone plugging gracefully, we can get rid of this override now. Reported-and-tested-by: Eric Shattow Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 766e56754c64..92b9b4324372 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3233,7 +3233,7 @@ static const struct hda_fixup stac927x_fixups[] = { /* configure the analog microphone on some laptops */ { 0x0c, 0x90a79130 }, /* correct the front output jack as a hp out */ - { 0x0f, 0x0227011f }, + { 0x0f, 0x0221101f }, /* correct the front input jack as a mic */ { 0x0e, 0x02a79130 }, {} -- cgit v1.2.3-70-g09d2 From e6058aaadcd473e5827720dc143af56aabbeecc7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:47:10 +0100 Subject: ASoC: jack: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5cccd7766..7aa26b5178aa 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -263,7 +263,7 @@ static irqreturn_t gpio_handler(int irq, void *data) if (device_may_wakeup(dev)) pm_wakeup_event(dev, gpio->debounce_time + 50); - schedule_delayed_work(&gpio->work, + queue_delayed_work(system_power_efficient_wq, &gpio->work, msecs_to_jiffies(gpio->debounce_time)); return IRQ_HANDLED; -- cgit v1.2.3-70-g09d2 From 8ccbc3ebe9a74f22c4f0acb363962f4e7c99d3cf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 15 Jul 2013 17:04:47 +0100 Subject: ASoC: mxs: Enable COMPILE_TEST builds Since DT based boards don't have any dependency on arch/arm enable them if COMPILE_TEST is enabled. Signed-off-by: Mark Brown --- sound/soc/mxs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 78d321cbe8b4..7daf860a2799 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,6 +1,6 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" - depends on ARCH_MXS + depends on ARCH_MXS || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- cgit v1.2.3-70-g09d2 From 2df7c6aad63f432befe51ac3144a96b37fa5b4ba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:43:00 +0100 Subject: ASoC: max98090: Use power efficient workqueue None of the delayed work the driver schedules has particularly short delays and it is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/max98090.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ad5313f98f28..0569a4c3ae00 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2084,8 +2084,9 @@ static irqreturn_t max98090_interrupt(int irq, void *data) pm_wakeup_event(codec->dev, 100); - schedule_delayed_work(&max98090->jack_work, - msecs_to_jiffies(100)); + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); } if (active & M98090_DRCACT_MASK) @@ -2132,8 +2133,9 @@ int max98090_mic_detect(struct snd_soc_codec *codec, snd_soc_jack_report(max98090->jack, 0, SND_JACK_HEADSET | SND_JACK_BTN_0); - schedule_delayed_work(&max98090->jack_work, - msecs_to_jiffies(100)); + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); return 0; } -- cgit v1.2.3-70-g09d2 From a14d982962c1c3caee99ddeea632d97fc851ea60 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:43:19 +0100 Subject: ASoC: sta32x: Use power efficient workqueue None of the delayed work the driver schedules has particularly short delays and it is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/sta32x.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index cfb55fe35e98..06edb396e733 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -363,16 +363,18 @@ static void sta32x_watchdog(struct work_struct *work) } if (!sta32x->shutdown) - schedule_delayed_work(&sta32x->watchdog_work, - round_jiffies_relative(HZ)); + queue_delayed_work(system_power_efficient_wq, + &sta32x->watchdog_work, + round_jiffies_relative(HZ)); } static void sta32x_watchdog_start(struct sta32x_priv *sta32x) { if (sta32x->pdata->needs_esd_watchdog) { sta32x->shutdown = 0; - schedule_delayed_work(&sta32x->watchdog_work, - round_jiffies_relative(HZ)); + queue_delayed_work(system_power_efficient_wq, + &sta32x->watchdog_work, + round_jiffies_relative(HZ)); } } -- cgit v1.2.3-70-g09d2 From 76394509f579cd4292b076f708da49404be6af37 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:44:03 +0100 Subject: ASoC: twl6040: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/twl6040.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 44621ddc332d..caf8784e7716 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -429,7 +429,8 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hs_jack.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- cgit v1.2.3-70-g09d2 From 2c5920a787e68e0a61886d24b05cdbde344c4b0c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:45:40 +0100 Subject: ASoC: wm8350: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/wm8350.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 0e8b3aaf6c8d..af1318ddb062 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1301,7 +1301,8 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hpl.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -1318,7 +1319,8 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hpr.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- cgit v1.2.3-70-g09d2 From ec1d648d6c6589986072913a7d45b1cef49eb4b0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:46:24 +0100 Subject: ASoC: wm8753: Use power efficient workqueue The work used to allow the capcitors to ramp is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/wm8753.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0a4ab4c423d1..d96ebf52d953 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1456,8 +1456,9 @@ static int wm8753_resume(struct snd_soc_codec *codec) if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); codec->dapm.bias_level = SND_SOC_BIAS_ON; - schedule_delayed_work(&codec->dapm.delayed_work, - msecs_to_jiffies(caps_charge)); + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, + msecs_to_jiffies(caps_charge)); } return 0; -- cgit v1.2.3-70-g09d2 From 68defe585f333223f0f3733340136d1b02003062 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:47:02 +0100 Subject: ASoC: wm8994: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/wm8994.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 02c320f71cdf..24131a7f9390 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -819,8 +819,9 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, * don't want false reports. */ if (wm8994->jackdet && !wm8994->clk_has_run) { - schedule_delayed_work(&wm8994->jackdet_bootstrap, - msecs_to_jiffies(1000)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->jackdet_bootstrap, + msecs_to_jiffies(1000)); wm8994->clk_has_run = true; } break; @@ -3487,7 +3488,8 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) pm_wakeup_event(codec->dev, 300); - schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250)); + queue_delayed_work(system_power_efficient_wq, + &priv->mic_work, msecs_to_jiffies(250)); return IRQ_HANDLED; } @@ -3575,8 +3577,9 @@ static void wm8958_mic_id(void *data, u16 status) /* If nothing present then clear our statuses */ dev_dbg(codec->dev, "Detected open circuit\n"); - schedule_delayed_work(&wm8994->open_circuit_work, - msecs_to_jiffies(2500)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->open_circuit_work, + msecs_to_jiffies(2500)); return; } @@ -3690,8 +3693,9 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) WM1811_JACKDET_DB, 0); delay = control->pdata.micdet_delay; - schedule_delayed_work(&wm8994->mic_work, - msecs_to_jiffies(delay)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->mic_work, + msecs_to_jiffies(delay)); } else { dev_dbg(codec->dev, "Jack not detected\n"); @@ -3940,8 +3944,9 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) id_delay = wm8994->wm8994->pdata.mic_id_delay; if (wm8994->mic_detecting) - schedule_delayed_work(&wm8994->mic_complete_work, - msecs_to_jiffies(id_delay)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->mic_complete_work, + msecs_to_jiffies(id_delay)); else wm8958_button_det(codec, reg); -- cgit v1.2.3-70-g09d2 From 0a9eaa39db136aaf998d3aa0f7f25c331def336a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 19 Jul 2013 11:40:13 +0100 Subject: ASoC: fsl_ssi: Provide register I/O functions by default Use the ARM version by default as that's the more generally portable one, it doesn't matter if they work well on random platforms when the goal is only build coverage. Signed-off-by: Mark Brown Acked-by: Timur Tabi --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c9974a4ac042..e12a9977a1a9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -36,7 +36,7 @@ #define read_ssi(addr) in_be32(addr) #define write_ssi(val, addr) out_be32(addr, val) #define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set) -#elif defined ARM +#else #define read_ssi(addr) readl(addr) #define write_ssi(val, addr) writel(val, addr) /* -- cgit v1.2.3-70-g09d2 From 444fc4b369f9341d2cbcffe2d1ffde4cad5b4945 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 19 Jul 2013 10:22:21 -0300 Subject: ASoC: wm8962: Do not call configure_bclk() inside wm8962_set_dai_sysclk() Currently after playing any audio file, we get the following error message: $ aplay clarinet.wav Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo $ wm8962 0-001a: Unsupported sysclk ratio 544 This error message appears about 5 seconds after the audio playback has finished. Quoting Mark Brown [1]: "The issue here is triggered by the machine switching from the FLL to direct MCLK usage where the MCLK isn't generating a useful ratio. I suspect we should just kill the configure_bclk() in set_sysclk(), that one isn't safe as we can't reconfigure a live SYSCLK and it's probably the one that generates your warnings." Confirmed that the "Unsupported sysclk ratio" error message comes from wm8962_set_dai_sysclk(), so get rid of wm8962_configure_bclk() inside this function. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2013-July/064241.html Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e2de9ecfd641..8b8905dfc510 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2621,8 +2621,6 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, wm8962->sysclk_rate = freq; - wm8962_configure_bclk(codec); - return 0; } -- cgit v1.2.3-70-g09d2 From 52f19b14ec18f3166e43cda6a16bb39ffb376053 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Fri, 19 Jul 2013 17:42:57 +0800 Subject: ASoC: atmel: add wm8904 based audio machine driver Add wm8904 based audio machine driver for Atmel EK board The following ek board based on it - at91sam9n12ek - sama5d3xek (d31, d33, d34, d35) Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/atmel-wm8904.txt | 55 +++++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_wm8904.c | 254 +++++++++++++++++++++ 4 files changed, 321 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt create mode 100644 sound/soc/atmel/atmel_wm8904.c (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt new file mode 100644 index 000000000000..8bbe50c884b6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt @@ -0,0 +1,55 @@ +Atmel ASoC driver with wm8904 audio codec complex + +Required properties: + - compatible: "atmel,asoc-wm8904" + - atmel,model: The user-visible name of this sound complex. + - atmel,audio-routing: A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the WM8904's pins, and the jacks on the board: + + WM8904 pins: + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + + Board connectors: + + * Headphone Jack + * Line In Jack + * Mic + + - atmel,ssc-controller: The phandle of the SSC controller + - atmel,audio-codec: The phandle of the WM8904 audio codec + +Optional properties: + - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: +sound { + compatible = "atmel,asoc-wm8904"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + + atmel,model = "wm8904 @ AT91SAM9N12EK"; + + atmel,audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "IN2L", "Line In Jack", + "IN2R", "Line In Jack", + "Mic", "MICBIAS", + "IN1L", "Mic"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8904>; +}; diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 1c0b1858638c..986323b4caad 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -33,6 +33,16 @@ config SND_AT91_SOC_SAM9G20_WM8731 Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. +config SND_ATMEL_SOC_WM8904 + tristate "Atmel ASoC driver for boards using WM8904 codec" + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_ATMEL_SOC_DMA + select SND_SOC_WM8904 + help + Say Y if you want to add support for Atmel ASoC driver for boards using + WM8904 codec. + config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 41967ccb6f41..922d4da57109 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -11,6 +11,8 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o +snd-atmel-soc-wm8904-objs := atmel_wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c new file mode 100644 index 000000000000..7222380131ea --- /dev/null +++ b/sound/soc/atmel/atmel_wm8904.c @@ -0,0 +1,254 @@ +/* + * atmel_wm8904 - Atmel ASoC driver for boards with WM8904 codec. + * + * Copyright (C) 2012 Atmel + * + * Author: Bo Shen + * + * GPLv2 or later + */ + +#include +#include +#include +#include +#include + +#include + +#include "../codecs/wm8904.h" +#include "atmel_ssc_dai.h" + +#define MCLK_RATE 32768 + +static struct clk *mclk; + +static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), +}; + +static int atmel_asoc_wm8904_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK, + 32768, params_rate(params) * 256); + if (ret < 0) { + pr_err("%s - failed to set wm8904 codec PLL.", __func__); + return ret; + } + + /* + * As here wm8904 use FLL output as its system clock + * so calling set_sysclk won't care freq parameter + * then we pass 0 + */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8904_CLK_FLL, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("%s -failed to set wm8904 SYSCLK\n", __func__); + return ret; + } + + return 0; +} + +static struct snd_soc_ops atmel_asoc_wm8904_ops = { + .hw_params = atmel_asoc_wm8904_hw_params, +}; + +static int atmel_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + switch (level) { + case SND_SOC_BIAS_PREPARE: + clk_prepare_enable(mclk); + break; + case SND_SOC_BIAS_OFF: + clk_disable_unprepare(mclk); + break; + default: + break; + } + } + + return 0; +}; + +static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { + .name = "WM8904", + .stream_name = "WM8904 PCM", + .codec_dai_name = "wm8904-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &atmel_asoc_wm8904_ops, +}; + +static struct snd_soc_card atmel_asoc_wm8904_card = { + .name = "atmel_asoc_wm8904", + .owner = THIS_MODULE, + .set_bias_level = atmel_set_bias_level, + .dai_link = &atmel_asoc_wm8904_dailink, + .num_links = 1, + .dapm_widgets = atmel_asoc_wm8904_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(atmel_asoc_wm8904_dapm_widgets), + .fully_routed = true, +}; + +static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *codec_np, *cpu_np; + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int ret; + + if (!np) { + dev_err(&pdev->dev, "only device tree supported\n"); + return -EINVAL; + } + + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) { + dev_err(&pdev->dev, "failed to parse card name\n"); + return ret; + } + + ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio routing\n"); + return ret; + } + + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "failed to get dai and pcm info\n"); + ret = -EINVAL; + return ret; + } + dailink->cpu_of_node = cpu_np; + dailink->platform_of_node = cpu_np; + of_node_put(cpu_np); + + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "failed to get codec info\n"); + ret = -EINVAL; + return ret; + } + dailink->codec_of_node = codec_np; + of_node_put(codec_np); + + return 0; +} + +static int atmel_asoc_wm8904_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + struct clk *clk_src; + struct pinctrl *pinctrl; + int id, ret; + + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + dev_err(&pdev->dev, "failed to request pinctrl\n"); + return PTR_ERR(pinctrl); + } + + card->dev = &pdev->dev; + ret = atmel_asoc_wm8904_dt_init(pdev); + if (ret) { + dev_err(&pdev->dev, "failed to init dt info\n"); + return ret; + } + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + ret = atmel_ssc_set_audio(id); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set SSC %d for audio\n", id); + return ret; + } + + mclk = clk_get(NULL, "pck0"); + if (IS_ERR(mclk)) { + dev_err(&pdev->dev, "failed to get pck0\n"); + ret = PTR_ERR(mclk); + goto err_set_audio; + } + + clk_src = clk_get(NULL, "clk32k"); + if (IS_ERR(clk_src)) { + dev_err(&pdev->dev, "failed to get clk32k\n"); + ret = PTR_ERR(clk_src); + goto err_set_audio; + } + + ret = clk_set_parent(mclk, clk_src); + clk_put(clk_src); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set MCLK parent\n"); + goto err_set_audio; + } + + dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE); + clk_set_rate(mclk, MCLK_RATE); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + goto err_set_audio; + } + + return 0; + +err_set_audio: + atmel_ssc_put_audio(id); + return ret; +} + +static int atmel_asoc_wm8904_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int id; + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + + snd_soc_unregister_card(card); + atmel_ssc_put_audio(id); + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = { + { .compatible = "atmel,asoc-wm8904", }, + { } +}; +#endif + +static struct platform_driver atmel_asoc_wm8904_driver = { + .driver = { + .name = "atmel-wm8904-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids), + }, + .probe = atmel_asoc_wm8904_probe, + .remove = atmel_asoc_wm8904_remove, +}; + +module_platform_driver(atmel_asoc_wm8904_driver); + +/* Module information */ +MODULE_AUTHOR("Bo Shen "); +MODULE_DESCRIPTION("ALSA SoC machine driver for Atmel EK with WM8904 codec"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From 83e2e4eeb85fd45ff592b79ea11a19df49df872e Mon Sep 17 00:00:00 2001 From: H Hartley Sweeten Date: Fri, 19 Jul 2013 09:53:25 -0700 Subject: ASoC: ep93xx: fix build of ep93xx-ac97.c Fix the build of this driver. It was broken by: Commit 453807f3006757a5661c4000262d7d9284b5214c ASoC: ep93xx: Use ep93xx_dma_params instead of ep93xx_pcm_dma_params The removed struct ep93xx_pcm_dma_params use the member 'dma_port' to select the dma channel. The struct ep93xx_dma_data uses the member 'port'. Signed-off-by: H Hartley Sweeten Cc: Ryan Mallon Cc: Lars-Peter Clausen Cc: Mark Brown Cc: Liam Girdwood Cc: Jaroslav Kysela Cc: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-ac97.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index ac73c607410a..04491f0e8d1b 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -102,13 +102,13 @@ static struct ep93xx_ac97_info *ep93xx_ac97_info; static struct ep93xx_dma_data ep93xx_ac97_pcm_out = { .name = "ac97-pcm-out", - .dma_port = EP93XX_DMA_AAC1, + .port = EP93XX_DMA_AAC1, .direction = DMA_MEM_TO_DEV, }; static struct ep93xx_dma_data ep93xx_ac97_pcm_in = { .name = "ac97-pcm-in", - .dma_port = EP93XX_DMA_AAC1, + .port = EP93XX_DMA_AAC1, .direction = DMA_DEV_TO_MEM, }; -- cgit v1.2.3-70-g09d2 From be2f93a4c4981b3646b6f98f477154411b8516cb Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Fri, 19 Jul 2013 18:26:53 +0200 Subject: ALSA: usb-audio: 6fire: return correct XRUN indication Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of 6fire, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Signed-off-by: Eldad Zack Cc: Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 2aa4e13063a8..3d2551cc10f2 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -543,7 +543,7 @@ static snd_pcm_uframes_t usb6fire_pcm_pointer( snd_pcm_uframes_t ret; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); ret = sub->dma_off; -- cgit v1.2.3-70-g09d2 From f3142807fdb965a7ae1c3a8a6fd91ff92a8efa7a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sat, 20 Jul 2013 16:16:01 -0300 Subject: ASoC: fsl: fsl_ssi: Add MODULE_ALIAS Add MODULE_ALIAS, so that auto module loading can work. Signed-off-by: Fabio Estevam Acked-by: Timur Tavi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e12a9977a1a9..11469fe773e2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,6 +900,7 @@ static struct platform_driver fsl_ssi_driver = { module_platform_driver(fsl_ssi_driver); +MODULE_ALIAS("platform:fsl-ssi-dai"); MODULE_AUTHOR("Timur Tabi "); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-70-g09d2 From 6e20b0d760759fd8c90a7b6ccfd6662e3edb94ed Mon Sep 17 00:00:00 2001 From: Michael Trimarchi Date: Sun, 21 Jul 2013 18:24:01 +0200 Subject: ASoC: omap-mcbsp: Support SND_SOC_DAIFMT_CBM_CFS for omap3/4 Add SND_SOC_DAIFMT_CBM_CFS support for omap3/omap4. The patch was tested on a pandaboard-es board connected to the pcm1792a codec. mcbspx_fsx must configured as output and mcbspx_clkx must be configured as input. Signed-off-by: Michael Trimarchi Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7483efb6dc67..6c19bba23570 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -433,6 +433,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* Sample rate generator drives the FS */ regs->srgr2 |= FSGM; break; + case SND_SOC_DAIFMT_CBM_CFS: + /* McBSP slave. FS clock as output */ + regs->srgr2 |= FSGM; + regs->pcr0 |= FSXM; + break; case SND_SOC_DAIFMT_CBM_CFM: /* McBSP slave */ break; -- cgit v1.2.3-70-g09d2 From 72192366f4e1385fe6e44600aa5b75d0136e3d52 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sun, 21 Jul 2013 13:39:08 -0300 Subject: ASoC: fsl: imx-audmux: Check the return value from clk_prepare_enable() clk_prepare_enable() may fail, so let's check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index e260f1f899db..1a5da1e13077 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -73,8 +73,11 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - if (audmux_clk) - clk_prepare_enable(audmux_clk); + if (audmux_clk) { + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; + } ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); @@ -224,14 +227,19 @@ EXPORT_SYMBOL_GPL(imx_audmux_v1_configure_port); int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, unsigned int pdcr) { + int ret; + if (audmux_type != IMX31_AUDMUX) return -EINVAL; if (!audmux_base) return -ENOSYS; - if (audmux_clk) - clk_prepare_enable(audmux_clk); + if (audmux_clk) { + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; + } writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); -- cgit v1.2.3-70-g09d2 From a06e427d088d8a9b81defd42e6bae5f1cd69fc3f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:44:03 +0100 Subject: ASoC: twl6040: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar Acked-by: Peter Ujfalusi --- sound/soc/codecs/twl6040.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 44621ddc332d..caf8784e7716 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -429,7 +429,8 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hs_jack.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- cgit v1.2.3-70-g09d2 From b5c745fb75b7e5ab06e9c99d63427595a234cc89 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 22 Jul 2013 09:56:54 +0300 Subject: ASoC: core: double free in snd_soc_add_platform() There are three callers for this function, and none of them want it to free platform for them. It leads to a double free. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0ec070cf7231..d82ee386eab5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3908,10 +3908,8 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, { /* create platform component name */ platform->name = fmt_single_name(dev, &platform->id); - if (platform->name == NULL) { - kfree(platform); + if (platform->name == NULL) return -ENOMEM; - } platform->dev = dev; platform->driver = platform_drv; -- cgit v1.2.3-70-g09d2 From 204f029155e7da98b59e6969cf29e210bbe84de5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 19 Jul 2013 12:10:18 +0100 Subject: ASoC: mxs: Remove unneeded mach-types.h inclusions Signed-off-by: Mark Brown Acked-by: Shawn Guo --- sound/soc/mxs/mxs-saif.c | 1 - sound/soc/mxs/mxs-sgtl5000.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 54511c5e6a7c..b56b8a0e8deb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -31,7 +31,6 @@ #include #include #include -#include #include "mxs-saif.h" diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 1b134d72f120..b2e372dd02eb 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -25,7 +25,6 @@ #include #include #include -#include #include "../codecs/sgtl5000.h" #include "mxs-saif.h" -- cgit v1.2.3-70-g09d2 From da72c9619f4df033d431a0a4cee715cf14c78433 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:46:46 +0100 Subject: ASoC: wm8962: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/wm8962.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8b8905dfc510..36782f067cc5 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3044,8 +3044,9 @@ static irqreturn_t wm8962_irq(int irq, void *data) pm_wakeup_event(dev, 300); - schedule_delayed_work(&wm8962->mic_work, - msecs_to_jiffies(250)); + queue_delayed_work(system_power_efficient_wq, + &wm8962->mic_work, + msecs_to_jiffies(250)); } return IRQ_HANDLED; -- cgit v1.2.3-70-g09d2 From 315d9c649f15413abebea2212bc5bc8915fc7d2a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 22 Jul 2013 12:17:31 +0100 Subject: ASoC: mxs: Depends on COMMON_CLK The SAIF driver is a clock provider so specifically needs the common clock implementedation. Reported-by: Fengguang Wu Signed-off-by: Mark Brown Acked-by: Shawn Guo --- sound/soc/mxs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 7daf860a2799..219235c02212 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" depends on ARCH_MXS || COMPILE_TEST + depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- cgit v1.2.3-70-g09d2 From 647ab784c507763bfda79155f125b6edd1244806 Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Sun, 21 Jul 2013 10:34:09 +0800 Subject: ASoC: tegra: correct playback_dma_data setup The errors were caused by copy/paste mistake in below commit since v3.10: 3489d50 ASoC: tegra: Use common DAI DMA data struct It also corrects slave_id initialization in tegra20_ac97 driver. Signed-off-by: Richard Zhao Acked-by: Stephen Warren Acked-by: Lucas Stach Signed-off-by: Mark Brown Cc: # 3.10 --- sound/soc/tegra/tegra20_ac97.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index e58233f7df61..6c486625321b 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -389,9 +389,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ac97->capture_dma_data.slave_id = of_dma[1]; ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1; - ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - ac97->capture_dma_data.maxburst = 4; - ac97->capture_dma_data.slave_id = of_dma[0]; + ac97->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + ac97->playback_dma_data.maxburst = 4; + ac97->playback_dma_data.slave_id = of_dma[1]; ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); if (ret) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 5eaa12cdc6eb..551b3c93ce93 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -323,8 +323,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) } spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; - spdif->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - spdif->capture_dma_data.maxburst = 4; + spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + spdif->playback_dma_data.maxburst = 4; spdif->playback_dma_data.slave_id = dmareq->start; pm_runtime_enable(&pdev->dev); -- cgit v1.2.3-70-g09d2 From 5f6e7d52c4959019d12a7deebbde548884a917d1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 23 Jul 2013 11:12:25 +0200 Subject: ASoC: Remove unused dapm_get_snd_card() and dapm_get_soc_card() These two functions were added two years ago in commit 4805608 ("ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.") but have remained unused so far. Considering that the dapm context actually has a direct pointer to the card the functions also seem to be unnecessary. E.g. the expressions 'dapm_get_soc_card(dapm)' and 'dapm->card' yield the same result. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 30 ------------------------------ 1 file changed, 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b94190820e8c..93ea5d9fe356 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -174,36 +174,6 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } -/* get snd_card from DAPM context */ -static inline struct snd_card *dapm_get_snd_card( - struct snd_soc_dapm_context *dapm) -{ - if (dapm->codec) - return dapm->codec->card->snd_card; - else if (dapm->platform) - return dapm->platform->card->snd_card; - else - BUG(); - - /* unreachable */ - return NULL; -} - -/* get soc_card from DAPM context */ -static inline struct snd_soc_card *dapm_get_soc_card( - struct snd_soc_dapm_context *dapm) -{ - if (dapm->codec) - return dapm->codec->card; - else if (dapm->platform) - return dapm->platform->card; - else - BUG(); - - /* unreachable */ - return NULL; -} - static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; -- cgit v1.2.3-70-g09d2 From af2d8b5d95f12c36bfabe90c0879923efedefd2c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 22 Jul 2013 18:49:52 +0200 Subject: ASoC: Build adau1701 when SND_SOC_ALL_CODECS is selected The adau1701 driver was removed from SND_SOC_ALL_CODECS in commit commit ee8c7e9 ("ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency") due to the dependency on the SigmaDSP firmware loader which was only available on a limited set of platforms. This was fixed quite some time ago in commit 40216ce7 ("ASoC: Move SigmaDSP firmware loader to ASoC") though, so we can add the driver back again to SND_SOC_ALL_CODECS. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..adddb39f3c33 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -20,6 +20,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI + select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C -- cgit v1.2.3-70-g09d2 From 96b9bc6174a030691a4a60b0117ba7718d2bb27a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 22 Jul 2013 18:49:53 +0200 Subject: ASoC: adau1701: Add adau1702 and adau1401(a) device ids Both the adau1702 and the adau1401(a) are register compatible to the adau1701, so add them to adau1701 driver's device id table. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d1124a5b3471..44d8a95d9ec3 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -734,7 +734,10 @@ static int adau1701_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id adau1701_i2c_id[] = { + { "adau1401", 0 }, + { "adau1401a", 0 }, { "adau1701", 0 }, + { "adau1702", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); -- cgit v1.2.3-70-g09d2 From fee4b700a4e9e446151eb5a03874ca8666323113 Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Tue, 23 Jul 2013 11:15:06 +0200 Subject: ALSA: hiface: return correct XRUN indication Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of hiface, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Cc: Antonio Ospite Signed-off-by: Eldad Zack Cc: Signed-off-by: Takashi Iwai --- sound/usb/hiface/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 6430ed2a9f65..c21a3df9a0df 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -503,7 +503,7 @@ static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) snd_pcm_uframes_t dma_offset; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); dma_offset = sub->dma_off; -- cgit v1.2.3-70-g09d2 From 56a678344273fd63f8ade26876283a2586a9bf3a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 24 Jul 2013 15:27:35 +0200 Subject: ASoC: dapm: Fix return value of snd_soc_dapm_put_{volsw,enum_virt}() The ALSA core expect the put callback of a control to return 1 if the value of the control changed and 0 if it did not. Both snd_soc_dapm_put_volsw() and snd_soc_dapm_put_enum_virt() currently always returns 0. For both functions we already have a 'change' variable which either contains 1 or 0 depending on whether the value has changed or not, so just return that. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b94190820e8c..bd16010441cc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2733,7 +2733,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, } mutex_unlock(&card->dapm_mutex); - return 0; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2861,7 +2861,6 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; - int ret = 0; int wi; if (ucontrol->value.enumerated.item[0] >= e->max) @@ -2881,7 +2880,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, } mutex_unlock(&card->dapm_mutex); - return ret; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); -- cgit v1.2.3-70-g09d2 From 63c69a6e4134a2085d40e40c02a395dd1bd8c023 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:03:01 +0100 Subject: ASoC: dapm: Use generic power check for everything except DAIs As noticed by Lars-Peter Clausen since the move to using widgets to hook into the DAIs we no longer directly manage the power of AIF or DAC/ADC widgets from the stream integration so they can just use the generic power checks instead of the custom stream integration ones they currently do. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bd16010441cc..378655839f74 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3126,16 +3126,16 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_value_mux: w->power_check = dapm_generic_check_power; break; - case snd_soc_dapm_adc: - case snd_soc_dapm_aif_out: case snd_soc_dapm_dai_out: w->power_check = dapm_adc_check_power; break; - case snd_soc_dapm_dac: - case snd_soc_dapm_aif_in: case snd_soc_dapm_dai_in: w->power_check = dapm_dac_check_power; break; + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: case snd_soc_dapm_input: -- cgit v1.2.3-70-g09d2 From c3f48ae6fd5a1ebdcaff5efe35f88f31daaee225 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 24 Jul 2013 15:27:36 +0200 Subject: ASoC: dapm: Pass snd_soc_card directly to soc_dpcm_runtime_update() soc_dpcm_runtime_update() operates on a ASoC card as a whole. Currently it takes a snd_soc_dapm_widget as its only parameter though. The widget is then used to look up the card and is otherwise unused. This patch changes the function to take a pointer to the card directly. This makes it possible to to call soc_dpcm_runtime_update() for updates which are not related to one specific widget. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dpcm.h | 2 +- sound/soc/soc-dapm.c | 4 ++-- sound/soc/soc-pcm.c | 10 +--------- 3 files changed, 4 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 04598f1efd77..047d657c331c 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -133,6 +133,6 @@ void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream, /* internal use only */ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); -int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *); +int soc_dpcm_runtime_update(struct snd_soc_card *); #endif diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 378655839f74..8d9c09b266fd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1986,7 +1986,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, ret = soc_dapm_mux_update_power(widget, kcontrol, mux, e); mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(widget); + soc_dpcm_runtime_update(card); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); @@ -2032,7 +2032,7 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, ret = soc_dapm_mixer_update_power(widget, kcontrol, connect); mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(widget); + soc_dpcm_runtime_update(card); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b6c640332a17..5c2c66209808 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1832,18 +1832,10 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) /* Called by DAPM mixer/mux changes to update audio routing between PCMs and * any DAI links. */ -int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *widget) +int soc_dpcm_runtime_update(struct snd_soc_card *card) { - struct snd_soc_card *card; int i, old, new, paths; - if (widget->codec) - card = widget->codec->card; - else if (widget->platform) - card = widget->platform->card; - else - return -EINVAL; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dapm_widget_list *list; -- cgit v1.2.3-70-g09d2 From ce6cfaf1de136cd3e6ed7c0ed984be8d003a58c1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 24 Jul 2013 15:27:37 +0200 Subject: ASoC: dapm: Run widget updates for shared controls at the same time Currently when updating a control that is shared between multiple widgets the whole power-up/power-down sequence is being run once for each widget. The control register is updated during the first run, which means the CODEC internal routing is also updated for all widgets during this first run. The input and output paths for each widgets are only updated though during the respective run for that widget. This leads to a slight inconsistency between the CODEC's internal state and ASoC's state, which causes non optimal behavior in regard to click and pop avoidance. E.g. consider the following setup where two MUXs share the same control. +------+ A1 ------| | | MUX1 |----- C1 B1 ------| | +------+ | control ---+ | +------+ A2 ------| | | MUX2 |----- C2 B2 ------| | +------+ If the control is updated to switch the MUXs from input A to input B with the current code the power-up/power-down sequence will look like this: Run soc_dapm_mux_update_power for MUX1 Power-down A1 Update MUXing Power-up B1 Run soc_dapm_mux_update_power for MUX2 Power-down A2 (Update MUXing) Power-up B2 Note that the second 'Update Muxing' is a no-op, since the register was already updated. While the preferred order for avoiding pops and clicks should be: Run soc_dapm_mux_update_power for control Power-down A1 Power-down A2 Update MUXing Power-up B1 Power-up B2 This patch changes the behavior to the later by running the updates for all widgets that the control is attached to at the same time. The new code is also a bit simpler since callers of soc_dapm_{mux,muxer}_update_power don't have to loop over each widget anymore and neither do we need to keep track for which of the kcontrol's widgets the current update is. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 7 +- sound/soc/soc-dapm.c | 162 ++++++++++++++++++++--------------------------- 2 files changed, 71 insertions(+), 98 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3e479f4e15f5..3717ad089486 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -391,10 +391,10 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* external DAPM widget events */ -int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, struct snd_kcontrol *kcontrol, int connect); -int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e); +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); @@ -559,7 +559,6 @@ struct snd_soc_dapm_widget { }; struct snd_soc_dapm_update { - struct snd_soc_dapm_widget *widget; struct snd_kcontrol *kcontrol; int reg; int mask; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8d9c09b266fd..6f8a01bf6ca8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1456,34 +1456,45 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, static void dapm_widget_update(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_update *update = dapm->update; - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget_list *wlist; + struct snd_soc_dapm_widget *w = NULL; + unsigned int wi; int ret; if (!update) return; - w = update->widget; + wlist = snd_kcontrol_chip(update->kcontrol); - if (w->event && - (w->event_flags & SND_SOC_DAPM_PRE_REG)) { - ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG); - if (ret != 0) - dev_err(dapm->dev, "ASoC: %s DAPM pre-event failed: %d\n", - w->name, ret); + for (wi = 0; wi < wlist->num_widgets; wi++) { + w = wlist->widgets[wi]; + + if (w->event && (w->event_flags & SND_SOC_DAPM_PRE_REG)) { + ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG); + if (ret != 0) + dev_err(dapm->dev, "ASoC: %s DAPM pre-event failed: %d\n", + w->name, ret); + } } + if (!w) + return; + ret = soc_widget_update_bits_locked(w, update->reg, update->mask, update->val); if (ret < 0) dev_err(dapm->dev, "ASoC: %s DAPM update failed: %d\n", w->name, ret); - if (w->event && - (w->event_flags & SND_SOC_DAPM_POST_REG)) { - ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG); - if (ret != 0) - dev_err(dapm->dev, "ASoC: %s DAPM post-event failed: %d\n", - w->name, ret); + for (wi = 0; wi < wlist->num_widgets; wi++) { + w = wlist->widgets[wi]; + + if (w->event && (w->event_flags & SND_SOC_DAPM_POST_REG)) { + ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG); + if (ret != 0) + dev_err(dapm->dev, "ASoC: %s DAPM post-event failed: %d\n", + w->name, ret); + } } } @@ -1936,19 +1947,14 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) #endif /* test and update the power status of a mux widget */ -static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, +static int soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; - if (widget->id != snd_soc_dapm_mux && - widget->id != snd_soc_dapm_virt_mux && - widget->id != snd_soc_dapm_value_mux) - return -ENODEV; - /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->dapm->card->paths, list) { + list_for_each_entry(path, &dapm->card->paths, list) { if (path->kcontrol != kcontrol) continue; @@ -1966,24 +1972,23 @@ static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, "mux disconnection"); path->connect = 0; /* old connection must be powered down */ } + dapm_mark_dirty(path->sink, "mux change"); } - if (found) { - dapm_mark_dirty(widget, "mux change"); - dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); - } + if (found) + dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); return found; } -int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) { - struct snd_soc_card *card = widget->dapm->card; + struct snd_soc_card *card = dapm->card; int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = soc_dapm_mux_update_power(widget, kcontrol, mux, e); + ret = soc_dapm_mux_update_power(dapm, kcontrol, mux, e); mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -1992,19 +1997,14 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); /* test and update the power status of a mixer or switch widget */ -static int soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, +static int soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; int found = 0; - if (widget->id != snd_soc_dapm_mixer && - widget->id != snd_soc_dapm_mixer_named_ctl && - widget->id != snd_soc_dapm_switch) - return -ENODEV; - /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->dapm->card->paths, list) { + list_for_each_entry(path, &dapm->card->paths, list) { if (path->kcontrol != kcontrol) continue; @@ -2012,24 +2012,23 @@ static int soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, found = 1; path->connect = connect; dapm_mark_dirty(path->source, "mixer connection"); + dapm_mark_dirty(path->sink, "mixer update"); } - if (found) { - dapm_mark_dirty(widget, "mixer update"); - dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); - } + if (found) + dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); return found; } -int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int connect) +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, + struct snd_kcontrol *kcontrol, int connect) { - struct snd_soc_card *card = widget->dapm->card; + struct snd_soc_card *card = dapm->card; int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = soc_dapm_mixer_update_power(widget, kcontrol, connect); + ret = soc_dapm_mixer_update_power(dapm, kcontrol, connect); mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -2695,7 +2694,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, unsigned int val; int connect, change; struct snd_soc_dapm_update update; - int wi; if (snd_soc_volsw_is_stereo(mc)) dev_warn(widget->dapm->dev, @@ -2714,22 +2712,16 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { - for (wi = 0; wi < wlist->num_widgets; wi++) { - widget = wlist->widgets[wi]; - - widget->value = val; + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - update.kcontrol = kcontrol; - update.widget = widget; - update.reg = reg; - update.mask = mask; - update.val = val; - widget->dapm->update = &update; + widget->dapm->update = &update; - soc_dapm_mixer_update_power(widget, kcontrol, connect); + soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect); - widget->dapm->update = NULL; - } + widget->dapm->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2784,7 +2776,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; - int wi; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -2802,22 +2793,17 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, change = snd_soc_test_bits(widget->codec, e->reg, mask, val); if (change) { - for (wi = 0; wi < wlist->num_widgets; wi++) { - widget = wlist->widgets[wi]; + widget->value = val; - widget->value = val; + update.kcontrol = kcontrol; + update.reg = e->reg; + update.mask = mask; + update.val = val; + widget->dapm->update = &update; - update.kcontrol = kcontrol; - update.widget = widget; - update.reg = e->reg; - update.mask = mask; - update.val = val; - widget->dapm->update = &update; + soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); - soc_dapm_mux_update_power(widget, kcontrol, mux, e); - - widget->dapm->update = NULL; - } + widget->dapm->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2861,7 +2847,6 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; - int wi; if (ucontrol->value.enumerated.item[0] >= e->max) return -EINVAL; @@ -2870,13 +2855,8 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, change = widget->value != ucontrol->value.enumerated.item[0]; if (change) { - for (wi = 0; wi < wlist->num_widgets; wi++) { - widget = wlist->widgets[wi]; - - widget->value = ucontrol->value.enumerated.item[0]; - - soc_dapm_mux_update_power(widget, kcontrol, widget->value, e); - } + widget->value = ucontrol->value.enumerated.item[0]; + soc_dapm_mux_update_power(widget->dapm, kcontrol, widget->value, e); } mutex_unlock(&card->dapm_mutex); @@ -2949,7 +2929,6 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; - int wi; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -2967,22 +2946,17 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, change = snd_soc_test_bits(widget->codec, e->reg, mask, val); if (change) { - for (wi = 0; wi < wlist->num_widgets; wi++) { - widget = wlist->widgets[wi]; - - widget->value = val; + widget->value = val; - update.kcontrol = kcontrol; - update.widget = widget; - update.reg = e->reg; - update.mask = mask; - update.val = val; - widget->dapm->update = &update; + update.kcontrol = kcontrol; + update.reg = e->reg; + update.mask = mask; + update.val = val; + widget->dapm->update = &update; - soc_dapm_mux_update_power(widget, kcontrol, mux, e); + soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); - widget->dapm->update = NULL; - } + widget->dapm->update = NULL; } mutex_unlock(&card->dapm_mutex); -- cgit v1.2.3-70-g09d2 From 6b3fc03b3b614ced09df96ca60ab6f627d8c240c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 24 Jul 2013 15:27:38 +0200 Subject: ASoC: dapm: Add a update parameter to snd_soc_dapm_{mux,mixer}_update_power In order to avoid race conditions the assignment of dapm->update should happen while card->dapm_mutex is being held. To allow CODEC drivers to run a register update when using snd_soc_dapm_mux_update_power() or snd_soc_dapm_mixer_update_power() add a update parameter to these two functions. The update parameter will be assigned to dapm->update while card->dapm_mutex is locked. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 7 +++++-- sound/soc/soc-dapm.c | 10 ++++++++-- 2 files changed, 13 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3717ad089486..e77c6f5a8390 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -333,6 +333,7 @@ struct snd_soc_dapm_route; struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; +struct snd_soc_dapm_update; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -392,9 +393,11 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* external DAPM widget events */ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, - struct snd_kcontrol *kcontrol, int connect); + struct snd_kcontrol *kcontrol, int connect, + struct snd_soc_dapm_update *update); int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, - struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e); + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e, + struct snd_soc_dapm_update *update); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6f8a01bf6ca8..758761146a42 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1982,13 +1982,16 @@ static int soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, } int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, - struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e, + struct snd_soc_dapm_update *update) { struct snd_soc_card *card = dapm->card; int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + dapm->update = update; ret = soc_dapm_mux_update_power(dapm, kcontrol, mux, e); + dapm->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -2022,13 +2025,16 @@ static int soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, } int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, - struct snd_kcontrol *kcontrol, int connect) + struct snd_kcontrol *kcontrol, int connect, + struct snd_soc_dapm_update *update) { struct snd_soc_card *card = dapm->card; int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + dapm->update = update; ret = soc_dapm_mixer_update_power(dapm, kcontrol, connect); + dapm->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); -- cgit v1.2.3-70-g09d2 From 5d99d778495cb02eafd38292f462c4466fc7189f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 24 Jul 2013 15:27:39 +0200 Subject: ASoC: tlv320aic3x: Use snd_soc_dapm_mixer_update_power Use snd_soc_dapm_mixer_update_power() instead of reimplementing its functionality. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 49 ++++++++++++++---------------------------- 1 file changed, 16 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e5b926883131..1325c0c0df50 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -147,10 +147,9 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val_mask; - int ret; - struct snd_soc_dapm_path *path; - int found = 0; + unsigned short val; + struct snd_soc_dapm_update update; + int connect, change; val = (ucontrol->value.integer.value[0] & mask); @@ -158,42 +157,26 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, if (val) val = mask; + connect = !!val; + if (invert) val = mask - val; - val_mask = mask << shift; - val = val << shift; - - mutex_lock(&widget->codec->mutex); - if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) { - /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->dapm->card->paths, list) { - if (path->kcontrol != kcontrol) - continue; + mask <<= shift; + val <<= shift; - /* found, now check type */ - found = 1; - if (val) - /* new connection */ - path->connect = invert ? 0 : 1; - else - /* old connection must be powered down */ - path->connect = invert ? 1 : 0; + change = snd_soc_test_bits(widget->codec, val, mask, reg); + if (change) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - dapm_mark_dirty(path->source, "tlv320aic3x source"); - dapm_mark_dirty(path->sink, "tlv320aic3x sink"); - - break; - } + snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect, + &update); } - mutex_unlock(&widget->codec->mutex); - - if (found) - snd_soc_dapm_sync(widget->dapm); - - ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val); - return ret; + return change; } /* -- cgit v1.2.3-70-g09d2 From 60db923dad880f973d8e1aa7f654f2b928b34ad1 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 23 Jul 2013 20:01:54 +0200 Subject: ASoC: au1x: don't check resource with devm_ioremap_resource devm_ioremap_resource does sanity checks on the given resource. No need to duplicate this in the driver. Signed-off-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/au1x/psc-ac97.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index a822ab822bb7..986dcec79fa0 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -379,9 +379,6 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) mutex_init(&wd->lock); iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) - return -ENODEV; - wd->mmio = devm_ioremap_resource(&pdev->dev, iores); if (IS_ERR(wd->mmio)) return PTR_ERR(wd->mmio); -- cgit v1.2.3-70-g09d2 From bd23ee0c5e2af1d3cdef813039384abd400dcffc Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 23 Jul 2013 20:01:55 +0200 Subject: ASoC: ep93xx: don't check resource with devm_ioremap_resource devm_ioremap_resource does sanity checks on the given resource. No need to duplicate this in the driver. Signed-off-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-ac97.c | 3 --- sound/soc/cirrus/ep93xx-i2s.c | 3 --- 2 files changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index ac73c607410a..2a5cdaea0c40 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -363,9 +363,6 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) - return -ENODEV; - info->regs = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(info->regs)) return PTR_ERR(info->regs); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 17ad70bca9fe..f23f331e9a97 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -376,9 +376,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) - return -ENODEV; - info->regs = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(info->regs)) return PTR_ERR(info->regs); -- cgit v1.2.3-70-g09d2 From 916dd4130c9d045b8b50c16211b4d68ce147e50f Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 23 Jul 2013 20:01:56 +0200 Subject: ASoC: nuc900: don't check resource with devm_ioremap_resource devm_ioremap_resource does sanity checks on the given resource. No need to duplicate this in the driver. Signed-off-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-ac97.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index f4c2417a8730..8987bf987e58 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -333,9 +333,6 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) spin_lock_init(&nuc900_audio->lock); nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!nuc900_audio->res) - return ret; - nuc900_audio->mmio = devm_ioremap_resource(&pdev->dev, nuc900_audio->res); if (IS_ERR(nuc900_audio->mmio)) -- cgit v1.2.3-70-g09d2 From 0b4fa3374172c07691dcf568e72f63fb41e82561 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 23 Jul 2013 20:01:57 +0200 Subject: ASoC: pxa: don't check resource with devm_ioremap_resource devm_ioremap_resource does sanity checks on the given resource. No need to duplicate this in the driver. Signed-off-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/pxa/mmp-sspa.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 62142ce367c7..1605934d525e 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -430,9 +430,6 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) - return -ENOMEM; - priv->sspa->mmio_base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(priv->sspa->mmio_base)) return PTR_ERR(priv->sspa->mmio_base); -- cgit v1.2.3-70-g09d2 From 06c77ea65b867e9b7ced67636b9a2854e0ce8558 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 23 Jul 2013 20:01:59 +0200 Subject: ASoC: txx9: don't check resource with devm_ioremap_resource devm_ioremap_resource does sanity checks on the given resource. No need to duplicate this in the driver. Signed-off-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc-ac97.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 4bcce8a3cded..e0305a148568 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -184,9 +184,6 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (irq < 0) return irq; r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) - return -EBUSY; - drvdata->base = devm_ioremap_resource(&pdev->dev, r); if (IS_ERR(drvdata->base)) return PTR_ERR(drvdata->base); -- cgit v1.2.3-70-g09d2 From 32bd8cd25759411d3e11351db59be05446092f80 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 25 Jul 2013 17:41:41 +0800 Subject: ASoC: fsl: Set sdma peripheral type directly Let CPU DAI drivers set SDMA periperal type directly to support more dma types(SPDIF, ESAI) other than only two for SSI. This will easily allow some non-SSI drivers to use the imx-pcm-dma as well. Signed-off-by: Nicolin Chen Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/fsl/imx-pcm.h | 7 ++----- sound/soc/fsl/imx-ssi.c | 4 ++-- 3 files changed, 6 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 11469fe773e2..4d78df7d7f34 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -775,9 +775,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) "fsl,spba-bus"); imx_pcm_dma_params_init_data(&ssi_private->filter_data_tx, - dma_events[0], shared); + dma_events[0], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx, - dma_events[1], shared); + dma_events[1], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); } /* Initialize the the device_attribute structure */ diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index fd56cad43cd6..9136625a3460 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -22,14 +22,11 @@ static inline void imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, - int dma, bool shared) + int dma, enum sdma_peripheral_type peripheral_type) { dma_data->dma_request = dma; dma_data->priority = DMA_PRIO_HIGH; - if (shared) - dma_data->peripheral_type = IMX_DMATYPE_SSI_SP; - else - dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->peripheral_type = peripheral_type; } struct imx_pcm_fiq_params { diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f029e27366de..f58bcd85c07f 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -571,13 +571,13 @@ static int imx_ssi_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); if (res) { imx_pcm_dma_params_init_data(&ssi->filter_data_tx, res->start, - false); + IMX_DMATYPE_SSI); } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); if (res) { imx_pcm_dma_params_init_data(&ssi->filter_data_rx, res->start, - false); + IMX_DMATYPE_SSI); } platform_set_drvdata(pdev, ssi); -- cgit v1.2.3-70-g09d2 From b60be4aa40cff1ebfffc09f92b9007ac1fa24fb4 Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Fri, 26 Jul 2013 19:06:48 +0530 Subject: ASoC: Samsung: I2S: Modify driver to give more flexibility This patch modifies the i2s driver to give flexibility towards register handling. This is a pre requirement for enabling i2s support on Exynos5420. This patch modifies only the required registers as a pre-requirement to support on Exynos5420. Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/i2s-regs.h | 36 +++++++++++++++------------ sound/soc/samsung/i2s.c | 58 ++++++++++++++++++++++++++------------------ 2 files changed, 55 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h index c0e6d9a19efc..30513b7ede3a 100644 --- a/sound/soc/samsung/i2s-regs.h +++ b/sound/soc/samsung/i2s-regs.h @@ -95,22 +95,26 @@ #define MOD_RXONLY (1 << 8) #define MOD_TXRX (2 << 8) #define MOD_MASK (3 << 8) -#define MOD_LR_LLOW (0 << 7) -#define MOD_LR_RLOW (1 << 7) -#define MOD_SDF_IIS (0 << 5) -#define MOD_SDF_MSB (1 << 5) -#define MOD_SDF_LSB (2 << 5) -#define MOD_SDF_MASK (3 << 5) -#define MOD_RCLK_256FS (0 << 3) -#define MOD_RCLK_512FS (1 << 3) -#define MOD_RCLK_384FS (2 << 3) -#define MOD_RCLK_768FS (3 << 3) -#define MOD_RCLK_MASK (3 << 3) -#define MOD_BCLK_32FS (0 << 1) -#define MOD_BCLK_48FS (1 << 1) -#define MOD_BCLK_16FS (2 << 1) -#define MOD_BCLK_24FS (3 << 1) -#define MOD_BCLK_MASK (3 << 1) +#define MOD_LRP_SHIFT 7 +#define MOD_LR_LLOW 0 +#define MOD_LR_RLOW 1 +#define MOD_SDF_SHIFT 5 +#define MOD_SDF_IIS 0 +#define MOD_SDF_MSB 1 +#define MOD_SDF_LSB 2 +#define MOD_SDF_MASK 3 +#define MOD_RCLK_SHIFT 3 +#define MOD_RCLK_256FS 0 +#define MOD_RCLK_512FS 1 +#define MOD_RCLK_384FS 2 +#define MOD_RCLK_768FS 3 +#define MOD_RCLK_MASK 3 +#define MOD_BCLK_SHIFT 1 +#define MOD_BCLK_32FS 0 +#define MOD_BCLK_48FS 1 +#define MOD_BCLK_16FS 2 +#define MOD_BCLK_24FS 3 +#define MOD_BCLK_MASK 3 #define MOD_8BIT (1 << 0) #define MOD_CDCLKCON (1 << 12) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 7a1734697434..973735841a05 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -198,7 +198,8 @@ static inline bool is_manager(struct i2s_dai *i2s) /* Read RCLK of I2S (in multiples of LRCLK) */ static inline unsigned get_rfs(struct i2s_dai *i2s) { - u32 rfs = (readl(i2s->addr + I2SMOD) >> 3) & 0x3; + u32 rfs = (readl(i2s->addr + I2SMOD) >> MOD_RCLK_SHIFT); + rfs &= MOD_RCLK_MASK; switch (rfs) { case 3: return 768; @@ -212,21 +213,22 @@ static inline unsigned get_rfs(struct i2s_dai *i2s) static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) { u32 mod = readl(i2s->addr + I2SMOD); + int rfs_shift = MOD_RCLK_SHIFT; - mod &= ~MOD_RCLK_MASK; + mod &= ~(MOD_RCLK_MASK << rfs_shift); switch (rfs) { case 768: - mod |= MOD_RCLK_768FS; + mod |= (MOD_RCLK_768FS << rfs_shift); break; case 512: - mod |= MOD_RCLK_512FS; + mod |= (MOD_RCLK_512FS << rfs_shift); break; case 384: - mod |= MOD_RCLK_384FS; + mod |= (MOD_RCLK_384FS << rfs_shift); break; default: - mod |= MOD_RCLK_256FS; + mod |= (MOD_RCLK_256FS << rfs_shift); break; } @@ -236,7 +238,8 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) /* Read Bit-Clock of I2S (in multiples of LRCLK) */ static inline unsigned get_bfs(struct i2s_dai *i2s) { - u32 bfs = (readl(i2s->addr + I2SMOD) >> 1) & 0x3; + u32 bfs = readl(i2s->addr + I2SMOD) >> MOD_BCLK_SHIFT; + bfs &= MOD_BCLK_MASK; switch (bfs) { case 3: return 24; @@ -250,21 +253,22 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { u32 mod = readl(i2s->addr + I2SMOD); + int bfs_shift = MOD_BCLK_SHIFT; - mod &= ~MOD_BCLK_MASK; + mod &= ~(MOD_BCLK_MASK << bfs_shift); switch (bfs) { case 48: - mod |= MOD_BCLK_48FS; + mod |= (MOD_BCLK_48FS << bfs_shift); break; case 32: - mod |= MOD_BCLK_32FS; + mod |= (MOD_BCLK_32FS << bfs_shift); break; case 24: - mod |= MOD_BCLK_24FS; + mod |= (MOD_BCLK_24FS << bfs_shift); break; case 16: - mod |= MOD_BCLK_16FS; + mod |= (MOD_BCLK_16FS << bfs_shift); break; default: dev_err(&i2s->pdev->dev, "Wrong BCLK Divider!\n"); @@ -491,20 +495,25 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, { struct i2s_dai *i2s = to_info(dai); u32 mod = readl(i2s->addr + I2SMOD); + int lrp_shift = MOD_LRP_SHIFT, sdf_shift = MOD_SDF_SHIFT; + int sdf_mask, lrp_rlow; u32 tmp = 0; + sdf_mask = MOD_SDF_MASK << sdf_shift; + lrp_rlow = MOD_LR_RLOW << lrp_shift; + /* Format is priority */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: - tmp |= MOD_LR_RLOW; - tmp |= MOD_SDF_MSB; + tmp |= lrp_rlow; + tmp |= (MOD_SDF_MSB << sdf_shift); break; case SND_SOC_DAIFMT_LEFT_J: - tmp |= MOD_LR_RLOW; - tmp |= MOD_SDF_LSB; + tmp |= lrp_rlow; + tmp |= (MOD_SDF_LSB << sdf_shift); break; case SND_SOC_DAIFMT_I2S: - tmp |= MOD_SDF_IIS; + tmp |= (MOD_SDF_IIS << sdf_shift); break; default: dev_err(&i2s->pdev->dev, "Format not supported\n"); @@ -519,10 +528,10 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_NB_IF: - if (tmp & MOD_LR_RLOW) - tmp &= ~MOD_LR_RLOW; + if (tmp & lrp_rlow) + tmp &= ~lrp_rlow; else - tmp |= MOD_LR_RLOW; + tmp |= lrp_rlow; break; default: dev_err(&i2s->pdev->dev, "Polarity not supported\n"); @@ -544,15 +553,18 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } + /* + * Don't change the I2S mode if any controller is active on this + * channel. + */ if (any_active(i2s) && - ((mod & (MOD_SDF_MASK | MOD_LR_RLOW - | MOD_SLAVE)) != tmp)) { + ((mod & (sdf_mask | lrp_rlow | MOD_SLAVE)) != tmp)) { dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); return -EAGAIN; } - mod &= ~(MOD_SDF_MASK | MOD_LR_RLOW | MOD_SLAVE); + mod &= ~(sdf_mask | lrp_rlow | MOD_SLAVE); mod |= tmp; writel(mod, i2s->addr + I2SMOD); -- cgit v1.2.3-70-g09d2 From ba51cbb8206cdba789a1f65b06526bb20f51d594 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 25 Jul 2013 19:40:17 +0300 Subject: ASoC: adau1701: type bug with ADAU1707_CLKDIV_UNSET ADAU1707_CLKDIV_UNSET is always compared against an unsigned int and not an unsigned long. The current tests are always false. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 44d8a95d9ec3..2c102522bbbc 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -91,7 +91,7 @@ #define ADAU1701_OSCIPOW_OPD 0x04 #define ADAU1701_DACSET_DACINIT 1 -#define ADAU1707_CLKDIV_UNSET (-1UL) +#define ADAU1707_CLKDIV_UNSET (-1U) #define ADAU1701_FIRMWARE "adau1701.bin" -- cgit v1.2.3-70-g09d2 From 02bd90e86dc63728feebaf2b238684208ccb19eb Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Sun, 28 Jul 2013 20:06:15 +0530 Subject: ASoC: compress: use soc_xxx handlers for metadata the compress metadata handlers were wrongly named sst_xxx Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 06a8000aa07b..d22015074670 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -334,7 +334,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_set_metadata(struct snd_compr_stream *cstream, +static int soc_compr_set_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -347,7 +347,7 @@ static int sst_compr_set_metadata(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_get_metadata(struct snd_compr_stream *cstream, +static int soc_compr_get_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -364,8 +364,8 @@ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, - .set_metadata = sst_compr_set_metadata, - .get_metadata = sst_compr_get_metadata, + .set_metadata = soc_compr_set_metadata, + .get_metadata = soc_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, -- cgit v1.2.3-70-g09d2 From 07ccc0f4f190070aaba8fb587307f7fefad97981 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 28 Jul 2013 18:45:28 +0200 Subject: ASoC: lm4857: Use table based setup for DAPM and controls Let the ASoC core take care of registering the DAPM widget and routes as well as the controls. This makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm4857.c | 26 +++++++------------------- 1 file changed, 7 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 9f9f59573f72..5ea2ed053eb7 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -174,28 +174,9 @@ static const struct snd_soc_dapm_route lm4857_routes[] = { static int lm4857_probe(struct snd_soc_codec *codec) { struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; codec->control_data = lm4857->i2c; - ret = snd_soc_add_codec_controls(codec, lm4857_controls, - ARRAY_SIZE(lm4857_controls)); - if (ret) - return ret; - - ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets, - ARRAY_SIZE(lm4857_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, lm4857_routes, - ARRAY_SIZE(lm4857_routes)); - if (ret) - return ret; - - snd_soc_dapm_new_widgets(dapm); - return 0; } @@ -207,6 +188,13 @@ static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { .reg_word_size = sizeof(uint8_t), .reg_cache_default = lm4857_default_regs, .set_bias_level = lm4857_set_bias_level, + + .controls = lm4857_controls, + .num_controls = ARRAY_SIZE(lm4857_controls), + .dapm_widgets = lm4857_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm4857_dapm_widgets), + .dapm_routes = lm4857_routes, + .num_dapm_routes = ARRAY_SIZE(lm4857_routes), }; static int lm4857_i2c_probe(struct i2c_client *i2c, -- cgit v1.2.3-70-g09d2 From 9b2709687a81297bca53f98100b6f13cd5e05b44 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 28 Jul 2013 18:45:29 +0200 Subject: ASoC: lm4857: Convert to regmap Use regmap for IO for the lm4857 driver instead of open-coding the IO read/write functions. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm4857.c | 85 +++++++++++++++-------------------------------- 1 file changed, 26 insertions(+), 59 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 5ea2ed053eb7..0e5743ea79df 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include @@ -23,12 +24,15 @@ #include struct lm4857 { - struct i2c_client *i2c; + struct regmap *regmap; uint8_t mode; }; -static const uint8_t lm4857_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, +static const struct reg_default lm4857_default_regs[] = { + { 0x0, 0x00 }, + { 0x1, 0x00 }, + { 0x2, 0x00 }, + { 0x3, 0x00 }, }; /* The register offsets in the cache array */ @@ -42,39 +46,6 @@ static const uint8_t lm4857_default_regs[] = { #define LM4857_WAKEUP 5 #define LM4857_EPGAIN 4 -static int lm4857_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - uint8_t data; - int ret; - - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return ret; - - data = (reg << 6) | value; - ret = i2c_master_send(codec->control_data, &data, 1); - if (ret != 1) { - dev_err(codec->dev, "Failed to write register: %d\n", ret); - return ret; - } - - return 0; -} - -static unsigned int lm4857_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - unsigned int val; - int ret; - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret) - return -1; - - return val; -} - static int lm4857_get_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -96,7 +67,7 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol, lm4857->mode = value; if (codec->dapm.bias_level == SND_SOC_BIAS_ON) - snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, value + 6); + regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, value + 6); return 1; } @@ -108,10 +79,11 @@ static int lm4857_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, lm4857->mode + 6); + regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, + lm4857->mode + 6); break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, 0); + regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, 0); break; default: break; @@ -171,22 +143,7 @@ static const struct snd_soc_dapm_route lm4857_routes[] = { {"EP", NULL, "IN"}, }; -static int lm4857_probe(struct snd_soc_codec *codec) -{ - struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - - codec->control_data = lm4857->i2c; - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { - .write = lm4857_write, - .read = lm4857_read, - .probe = lm4857_probe, - .reg_cache_size = ARRAY_SIZE(lm4857_default_regs), - .reg_word_size = sizeof(uint8_t), - .reg_cache_default = lm4857_default_regs, .set_bias_level = lm4857_set_bias_level, .controls = lm4857_controls, @@ -197,11 +154,21 @@ static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { .num_dapm_routes = ARRAY_SIZE(lm4857_routes), }; +static const struct regmap_config lm4857_regmap_config = { + .val_bits = 6, + .reg_bits = 2, + + .max_register = LM4857_CTRL, + + .cache_type = REGCACHE_FLAT, + .reg_defaults = lm4857_default_regs, + .num_reg_defaults = ARRAY_SIZE(lm4857_default_regs), +}; + static int lm4857_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct lm4857 *lm4857; - int ret; lm4857 = devm_kzalloc(&i2c->dev, sizeof(*lm4857), GFP_KERNEL); if (!lm4857) @@ -209,11 +176,11 @@ static int lm4857_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, lm4857); - lm4857->i2c = i2c; - - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); + lm4857->regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config); + if (IS_ERR(lm4857->regmap)) + return PTR_ERR(lm4857->regmap); - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); } static int lm4857_i2c_remove(struct i2c_client *i2c) -- cgit v1.2.3-70-g09d2 From 1536a968892aa9095aada4b6d2ed326432cd71c8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 21 Jul 2013 21:35:52 -0700 Subject: ASoC: add Renesas R-Car core feature Renesas R-Car series sound circuit consists of SSI and its peripheral. But this peripheral circuits are different between R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2). (Actually, there are many difference in Generation1 chips) Basically, for the future, Renesas R-Car series will use Gen2 style sound circuit, but driver should care Gen1 also. The main differences between Gen1 and Gen2 peripheral are 1) register offset, 2) data path. This patch adds basic (core) feature for R-Car series sound driver as prototype Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 33 +++ sound/soc/sh/Kconfig | 7 + sound/soc/sh/Makefile | 3 + sound/soc/sh/rcar/Makefile | 2 + sound/soc/sh/rcar/core.c | 554 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 94 ++++++++ 6 files changed, 693 insertions(+) create mode 100644 include/sound/rcar_snd.h create mode 100644 sound/soc/sh/rcar/Makefile create mode 100644 sound/soc/sh/rcar/core.c create mode 100644 sound/soc/sh/rcar/rsnd.h (limited to 'sound') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h new file mode 100644 index 000000000000..7272b2ea7108 --- /dev/null +++ b/include/sound/rcar_snd.h @@ -0,0 +1,33 @@ +/* + * Renesas R-Car SRU/SCU/SSIU/SSI support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef RCAR_SND_H +#define RCAR_SND_H + +#include + + +#define RSND_BASE_MAX 0 + +struct rsnd_dai_platform_info { + int ssi_id_playback; + int ssi_id_capture; +}; + +struct rcar_snd_info { + u32 flags; + struct rsnd_dai_platform_info *dai_info; + int dai_info_nr; + int (*start)(int id); + int (*stop)(int id); +}; + +#endif diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 6bcb1164d599..56d8ff6a402d 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -34,6 +34,13 @@ config SND_SOC_SH4_SIU select SH_DMAE select FW_LOADER +config SND_SOC_RCAR + tristate "R-Car series SRU/SCU/SSIU/SSI support" + select SND_SIMPLE_CARD + select RCAR_CLK_ADG + help + This option enables R-Car SUR/SCU/SSIU/SSI sound support + ## ## Boards ## diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 849b387d17d9..aaf3dcd1ee2a 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -12,6 +12,9 @@ obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o +## audio units for R-Car +obj-$(CONFIG_SND_SOC_RCAR) += rcar/ + ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-migor-objs := migor.o diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile new file mode 100644 index 000000000000..cd8089f20c94 --- /dev/null +++ b/sound/soc/sh/rcar/Makefile @@ -0,0 +1,2 @@ +snd-soc-rcar-objs := core.o +obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c new file mode 100644 index 000000000000..13b5d50efd06 --- /dev/null +++ b/sound/soc/sh/rcar/core.c @@ -0,0 +1,554 @@ +/* + * Renesas R-Car SRU/SCU/SSIU/SSI support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto + * + * Based on fsi.c + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +/* + * Renesas R-Car sound device structure + * + * Gen1 + * + * SRU : Sound Routing Unit + * - SRC : Sampling Rate Converter + * - CMD + * - CTU : Channel Count Conversion Unit + * - MIX : Mixer + * - DVC : Digital Volume and Mute Function + * - SSI : Serial Sound Interface + * + * Gen2 + * + * SCU : Sampling Rate Converter Unit + * - SRC : Sampling Rate Converter + * - CMD + * - CTU : Channel Count Conversion Unit + * - MIX : Mixer + * - DVC : Digital Volume and Mute Function + * SSIU : Serial Sound Interface Unit + * - SSI : Serial Sound Interface + */ + +/* + * driver data Image + * + * rsnd_priv + * | + * | ** this depends on Gen1/Gen2 + * | + * +- gen + * | + * | ** these depend on data path + * | ** gen and platform data control it + * | + * +- rdai[0] + * | | sru ssiu ssi + * | +- playback -> [mod] -> [mod] -> [mod] -> ... + * | | + * | | sru ssiu ssi + * | +- capture -> [mod] -> [mod] -> [mod] -> ... + * | + * +- rdai[1] + * | | sru ssiu ssi + * | +- playback -> [mod] -> [mod] -> [mod] -> ... + * | | + * | | sru ssiu ssi + * | +- capture -> [mod] -> [mod] -> [mod] -> ... + * ... + * | + * | ** these control ssi + * | + * +- ssi + * | | + * | +- ssi[0] + * | +- ssi[1] + * | +- ssi[2] + * | ... + * | + * | ** these control scu + * | + * +- scu + * | + * +- scu[0] + * +- scu[1] + * +- scu[2] + * ... + * + * + * for_each_rsnd_dai(xx, priv, xx) + * rdai[0] => rdai[1] => rdai[2] => ... + * + * for_each_rsnd_mod(xx, rdai, xx) + * [mod] => [mod] => [mod] => ... + * + * rsnd_dai_call(xxx, fn ) + * [mod]->fn() -> [mod]->fn() -> [mod]->fn()... + * + */ +#include +#include "rsnd.h" + +#define RSND_RATES SNDRV_PCM_RATE_8000_96000 +#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) + +/* + * rsnd_platform functions + */ +#define rsnd_platform_call(priv, dai, func, param...) \ + (!(priv->info->func) ? -ENODEV : \ + priv->info->func(param)) + + +/* + * rsnd_dai functions + */ +struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id) +{ + return priv->rdai + id; +} + +static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) +{ + struct rsnd_priv *priv = snd_soc_dai_get_drvdata(dai); + + return rsnd_dai_get(priv, dai->id); +} + +int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io) +{ + return &rdai->playback == io; +} + +/* + * rsnd_soc_dai functions + */ +int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional) +{ + struct snd_pcm_substream *substream = io->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + int pos = io->byte_pos + additional; + + pos %= (runtime->periods * io->byte_per_period); + + return pos; +} + +void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int byte) +{ + io->byte_pos += byte; + + if (io->byte_pos >= io->next_period_byte) { + struct snd_pcm_substream *substream = io->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + + io->period_pos++; + io->next_period_byte += io->byte_per_period; + + if (io->period_pos >= runtime->periods) { + io->byte_pos = 0; + io->period_pos = 0; + io->next_period_byte = io->byte_per_period; + } + + snd_pcm_period_elapsed(substream); + } +} + +static int rsnd_dai_stream_init(struct rsnd_dai_stream *io, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + if (!list_empty(&io->head)) + return -EIO; + + INIT_LIST_HEAD(&io->head); + io->substream = substream; + io->byte_pos = 0; + io->period_pos = 0; + io->byte_per_period = runtime->period_size * + runtime->channels * + samples_to_bytes(runtime, 1); + io->next_period_byte = io->byte_per_period; + + return 0; +} + +static +struct snd_soc_dai *rsnd_substream_to_dai(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + return rtd->cpu_dai; +} + +static +struct rsnd_dai_stream *rsnd_rdai_to_io(struct rsnd_dai *rdai, + struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return &rdai->playback; + else + return &rdai->capture; +} + +static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct rsnd_priv *priv = snd_soc_dai_get_drvdata(dai); + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); + struct rsnd_dai_platform_info *info = rsnd_dai_get_platform_info(rdai); + int ssi_id = rsnd_dai_is_play(rdai, io) ? info->ssi_id_playback : + info->ssi_id_capture; + int ret; + unsigned long flags; + + rsnd_lock(priv, flags); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = rsnd_dai_stream_init(io, substream); + if (ret < 0) + goto dai_trigger_end; + + ret = rsnd_platform_call(priv, dai, start, ssi_id); + if (ret < 0) + goto dai_trigger_end; + + break; + case SNDRV_PCM_TRIGGER_STOP: + ret = rsnd_platform_call(priv, dai, stop, ssi_id); + if (ret < 0) + goto dai_trigger_end; + + break; + default: + ret = -EINVAL; + } + +dai_trigger_end: + rsnd_unlock(priv, flags); + + return ret; +} + +static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rdai->clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rdai->clk_master = 0; + break; + default: + return -EINVAL; + } + + /* set clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + rdai->bit_clk_inv = 0; + rdai->frm_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + rdai->bit_clk_inv = 1; + rdai->frm_clk_inv = 0; + break; + case SND_SOC_DAIFMT_IB_IF: + rdai->bit_clk_inv = 1; + rdai->frm_clk_inv = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + default: + rdai->bit_clk_inv = 0; + rdai->frm_clk_inv = 0; + break; + } + + /* set format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rdai->sys_delay = 0; + rdai->data_alignment = 0; + break; + case SND_SOC_DAIFMT_LEFT_J: + rdai->sys_delay = 1; + rdai->data_alignment = 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + rdai->sys_delay = 1; + rdai->data_alignment = 1; + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { + .trigger = rsnd_soc_dai_trigger, + .set_fmt = rsnd_soc_dai_set_fmt, +}; + +static int rsnd_dai_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct snd_soc_dai_driver *drv; + struct rsnd_dai *rdai; + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dai_platform_info *dai_info; + int dai_nr = info->dai_info_nr; + int i, pid, cid; + + drv = devm_kzalloc(dev, sizeof(*drv) * dai_nr, GFP_KERNEL); + rdai = devm_kzalloc(dev, sizeof(*rdai) * dai_nr, GFP_KERNEL); + if (!drv || !rdai) { + dev_err(dev, "dai allocate failed\n"); + return -ENOMEM; + } + + for (i = 0; i < dai_nr; i++) { + dai_info = &info->dai_info[i]; + + pid = dai_info->ssi_id_playback; + cid = dai_info->ssi_id_capture; + + /* + * init rsnd_dai + */ + INIT_LIST_HEAD(&rdai[i].playback.head); + INIT_LIST_HEAD(&rdai[i].capture.head); + + rdai[i].info = dai_info; + + snprintf(rdai[i].name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", i); + + /* + * init snd_soc_dai_driver + */ + drv[i].name = rdai[i].name; + drv[i].ops = &rsnd_soc_dai_ops; + if (pid >= 0) { + drv[i].playback.rates = RSND_RATES; + drv[i].playback.formats = RSND_FMTS; + drv[i].playback.channels_min = 2; + drv[i].playback.channels_max = 2; + } + if (cid >= 0) { + drv[i].capture.rates = RSND_RATES; + drv[i].capture.formats = RSND_FMTS; + drv[i].capture.channels_min = 2; + drv[i].capture.channels_max = 2; + } + + dev_dbg(dev, "%s (%d, %d) probed", rdai[i].name, pid, cid); + } + + priv->dai_nr = dai_nr; + priv->daidrv = drv; + priv->rdai = rdai; + + return 0; +} + +static void rsnd_dai_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ +} + +/* + * pcm ops + */ +static struct snd_pcm_hardware rsnd_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE, + .formats = RSND_FMTS, + .rates = RSND_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 256, +}; + +static int rsnd_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + + return ret; +} + +static int rsnd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_dai *dai = rsnd_substream_to_dai(substream); + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); + + return bytes_to_frames(runtime, io->byte_pos); +} + +static struct snd_pcm_ops rsnd_pcm_ops = { + .open = rsnd_pcm_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = rsnd_hw_params, + .hw_free = snd_pcm_lib_free_pages, + .pointer = rsnd_pointer, +}; + +/* + * snd_soc_platform + */ + +#define PREALLOC_BUFFER (32 * 1024) +#define PREALLOC_BUFFER_MAX (32 * 1024) + +static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + return snd_pcm_lib_preallocate_pages_for_all( + rtd->pcm, + SNDRV_DMA_TYPE_DEV, + rtd->card->snd_card->dev, + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); +} + +static void rsnd_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static struct snd_soc_platform_driver rsnd_soc_platform = { + .ops = &rsnd_pcm_ops, + .pcm_new = rsnd_pcm_new, + .pcm_free = rsnd_pcm_free, +}; + +static const struct snd_soc_component_driver rsnd_soc_component = { + .name = "rsnd", +}; + +/* + * rsnd probe + */ +static int rsnd_probe(struct platform_device *pdev) +{ + struct rcar_snd_info *info; + struct rsnd_priv *priv; + struct device *dev = &pdev->dev; + int ret; + + info = pdev->dev.platform_data; + if (!info) { + dev_err(dev, "driver needs R-Car sound information\n"); + return -ENODEV; + } + + /* + * init priv data + */ + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) { + dev_err(dev, "priv allocate failed\n"); + return -ENODEV; + } + + priv->dev = dev; + priv->info = info; + spin_lock_init(&priv->lock); + + /* + * init each module + */ + ret = rsnd_dai_probe(pdev, info, priv); + if (ret < 0) + return ret; + + /* + * asoc register + */ + ret = snd_soc_register_platform(dev, &rsnd_soc_platform); + if (ret < 0) { + dev_err(dev, "cannot snd soc register\n"); + return ret; + } + + ret = snd_soc_register_component(dev, &rsnd_soc_component, + priv->daidrv, rsnd_dai_nr(priv)); + if (ret < 0) { + dev_err(dev, "cannot snd dai register\n"); + goto exit_snd_soc; + } + + dev_set_drvdata(dev, priv); + + pm_runtime_enable(dev); + + dev_info(dev, "probed\n"); + return ret; + +exit_snd_soc: + snd_soc_unregister_platform(dev); + + return ret; +} + +static int rsnd_remove(struct platform_device *pdev) +{ + struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + + /* + * remove each module + */ + rsnd_dai_remove(pdev, priv); + + return 0; +} + +static struct platform_driver rsnd_driver = { + .driver = { + .name = "rcar_sound", + }, + .probe = rsnd_probe, + .remove = rsnd_remove, +}; +module_platform_driver(rsnd_driver); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Renesas R-Car audio driver"); +MODULE_AUTHOR("Kuninori Morimoto "); +MODULE_ALIAS("platform:rcar-pcm-audio"); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h new file mode 100644 index 000000000000..8d04fd0352bd --- /dev/null +++ b/sound/soc/sh/rcar/rsnd.h @@ -0,0 +1,94 @@ +/* + * Renesas R-Car + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#ifndef RSND_H +#define RSND_H + +#include +#include +#include +#include +#include +#include +#include +#include + +/* + * pseudo register + * + * The register address offsets SRU/SCU/SSIU on Gen1/Gen2 are very different. + * This driver uses pseudo register in order to hide it. + * see gen1/gen2 for detail + */ +struct rsnd_priv; +struct rsnd_dai; +struct rsnd_dai_stream; + +/* + * R-Car sound DAI + */ +#define RSND_DAI_NAME_SIZE 16 +struct rsnd_dai_stream { + struct list_head head; /* head of rsnd_mod list */ + struct snd_pcm_substream *substream; + int byte_pos; + int period_pos; + int byte_per_period; + int next_period_byte; +}; + +struct rsnd_dai { + char name[RSND_DAI_NAME_SIZE]; + struct rsnd_dai_platform_info *info; /* rcar_snd.h */ + struct rsnd_dai_stream playback; + struct rsnd_dai_stream capture; + + int clk_master:1; + int bit_clk_inv:1; + int frm_clk_inv:1; + int sys_delay:1; + int data_alignment:1; +}; + +#define rsnd_dai_nr(priv) ((priv)->dai_nr) +#define for_each_rsnd_dai(rdai, priv, i) \ + for (i = 0, (rdai) = rsnd_dai_get(priv, i); \ + i < rsnd_dai_nr(priv); \ + i++, (rdai) = rsnd_dai_get(priv, i)) + +struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id); +int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io); +#define rsnd_dai_get_platform_info(rdai) ((rdai)->info) + +void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int cnt); +int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional); + +/* + * R-Car sound priv + */ +struct rsnd_priv { + + struct device *dev; + struct rcar_snd_info *info; + spinlock_t lock; + + /* + * below value will be filled on rsnd_dai_probe() + */ + struct snd_soc_dai_driver *daidrv; + struct rsnd_dai *rdai; + int dai_nr; +}; + +#define rsnd_priv_to_dev(priv) ((priv)->dev) +#define rsnd_lock(priv, flags) spin_lock_irqsave(&priv->lock, flags) +#define rsnd_unlock(priv, flags) spin_unlock_irqrestore(&priv->lock, flags) + +#endif -- cgit v1.2.3-70-g09d2 From cdaa3cdfb4a710545a53740b1780a683b043618a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 21 Jul 2013 21:36:03 -0700 Subject: ASoC: add Renesas R-Car module feature Renesas R-Car series sound circuit consists of SSI and its peripheral. But this peripheral circuit is different between R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2) (Actually, there are many difference in Generation1 chips) Gen1 series consists of SRU/SSI/ADG, and Gen2 series consists of SCU/SSIU/SSI/ADG. In order to control these by same method, these are treated as "mod" on this driver. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 80 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 46 ++++++++++++++++++++++++++++ 2 files changed, 126 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 13b5d50efd06..a47fda2aa600 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -107,9 +107,74 @@ priv->info->func(param)) +/* + * rsnd_mod functions + */ +char *rsnd_mod_name(struct rsnd_mod *mod) +{ + if (!mod || !mod->ops) + return "unknown"; + + return mod->ops->name; +} + +void rsnd_mod_init(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_mod_ops *ops, + int id) +{ + mod->priv = priv; + mod->id = id; + mod->ops = ops; + INIT_LIST_HEAD(&mod->list); +} + /* * rsnd_dai functions */ +#define rsnd_dai_call(rdai, io, fn) \ +({ \ + struct rsnd_mod *mod, *n; \ + int ret = 0; \ + for_each_rsnd_mod(mod, n, io) { \ + ret = rsnd_mod_call(mod, fn, rdai, io); \ + if (ret < 0) \ + break; \ + } \ + ret; \ +}) + +int rsnd_dai_connect(struct rsnd_dai *rdai, + struct rsnd_mod *mod, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + if (!mod) { + dev_err(dev, "NULL mod\n"); + return -EIO; + } + + if (!list_empty(&mod->list)) { + dev_err(dev, "%s%d is not empty\n", + rsnd_mod_name(mod), + rsnd_mod_id(mod)); + return -EIO; + } + + list_add_tail(&mod->list, &io->head); + + return 0; +} + +int rsnd_dai_disconnect(struct rsnd_mod *mod) +{ + list_del_init(&mod->list); + + return 0; +} + struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id) { return priv->rdai + id; @@ -224,8 +289,23 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, if (ret < 0) goto dai_trigger_end; + ret = rsnd_dai_call(rdai, io, init); + if (ret < 0) + goto dai_trigger_end; + + ret = rsnd_dai_call(rdai, io, start); + if (ret < 0) + goto dai_trigger_end; break; case SNDRV_PCM_TRIGGER_STOP: + ret = rsnd_dai_call(rdai, io, stop); + if (ret < 0) + goto dai_trigger_end; + + ret = rsnd_dai_call(rdai, io, quit); + if (ret < 0) + goto dai_trigger_end; + ret = rsnd_platform_call(priv, dai, stop, ssi_id); if (ret < 0) goto dai_trigger_end; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 8d04fd0352bd..65d3835cffbc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -28,9 +28,52 @@ * see gen1/gen2 for detail */ struct rsnd_priv; +struct rsnd_mod; struct rsnd_dai; struct rsnd_dai_stream; +/* + * R-Car sound mod + */ + +struct rsnd_mod_ops { + char *name; + int (*init)(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); + int (*quit)(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); + int (*start)(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); + int (*stop)(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); +}; + +struct rsnd_mod { + int id; + struct rsnd_priv *priv; + struct rsnd_mod_ops *ops; + struct list_head list; /* connect to rsnd_dai playback/capture */ +}; + +#define rsnd_mod_to_priv(mod) ((mod)->priv) +#define rsnd_mod_id(mod) ((mod)->id) +#define for_each_rsnd_mod(pos, n, io) \ + list_for_each_entry_safe(pos, n, &(io)->head, list) +#define rsnd_mod_call(mod, func, rdai, io) \ + (!(mod) ? -ENODEV : \ + !((mod)->ops->func) ? 0 : \ + (mod)->ops->func(mod, rdai, io)) + +void rsnd_mod_init(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_mod_ops *ops, + int id); +char *rsnd_mod_name(struct rsnd_mod *mod); + /* * R-Car sound DAI */ @@ -64,6 +107,9 @@ struct rsnd_dai { i++, (rdai) = rsnd_dai_get(priv, i)) struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id); +int rsnd_dai_disconnect(struct rsnd_mod *mod); +int rsnd_dai_connect(struct rsnd_dai *rdai, struct rsnd_mod *mod, + struct rsnd_dai_stream *io); int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io); #define rsnd_dai_get_platform_info(rdai) ((rdai)->info) -- cgit v1.2.3-70-g09d2 From 3337744ac41bee00b0068ad5f926dd9c27540809 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 21 Jul 2013 21:36:21 -0700 Subject: ASoC: add Renesas R-Car Generation feature Renesas R-Car series sound circuit consists of SSI and its peripheral. But this peripheral circuit is different between R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2) (Actually, there are many difference in Generation1 chips) The main difference between Gen1 and Gen2 are 1) register offset, 2) data path In order to control Gen1/Gen2 by same method, this patch adds gen.c. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 10 +++ sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 58 ++++++++++++++++- sound/soc/sh/rcar/gen.c | 154 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 47 ++++++++++++++ 5 files changed, 269 insertions(+), 2 deletions(-) create mode 100644 sound/soc/sh/rcar/gen.c (limited to 'sound') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index 7272b2ea7108..14942a827fe5 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -22,6 +22,16 @@ struct rsnd_dai_platform_info { int ssi_id_capture; }; +/* + * flags + * + * 0x0000000A + * + * A : generation + */ +#define RSND_GEN1 (1 << 0) /* fixme */ +#define RSND_GEN2 (2 << 0) /* fixme */ + struct rcar_snd_info { u32 flags; struct rsnd_dai_platform_info *dai_info; diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index cd8089f20c94..b2d313b1eb94 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,2 +1,2 @@ -snd-soc-rcar-objs := core.o +snd-soc-rcar-objs := core.o gen.o obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index a47fda2aa600..bb8959f93a7d 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -107,6 +107,50 @@ priv->info->func(param)) +/* + * basic function + */ +u32 rsnd_read(struct rsnd_priv *priv, + struct rsnd_mod *mod, enum rsnd_reg reg) +{ + void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); + + BUG_ON(!base); + + return ioread32(base); +} + +void rsnd_write(struct rsnd_priv *priv, + struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data) +{ + void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); + struct device *dev = rsnd_priv_to_dev(priv); + + BUG_ON(!base); + + dev_dbg(dev, "w %p : %08x\n", base, data); + + iowrite32(data, base); +} + +void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, + enum rsnd_reg reg, u32 mask, u32 data) +{ + void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); + struct device *dev = rsnd_priv_to_dev(priv); + u32 val; + + BUG_ON(!base); + + val = ioread32(base); + val &= ~mask; + val |= data & mask; + iowrite32(val, base); + + dev_dbg(dev, "s %p : %08x\n", base, val); +} + /* * rsnd_mod functions */ @@ -289,6 +333,10 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, if (ret < 0) goto dai_trigger_end; + ret = rsnd_gen_path_init(priv, rdai, io); + if (ret < 0) + goto dai_trigger_end; + ret = rsnd_dai_call(rdai, io, init); if (ret < 0) goto dai_trigger_end; @@ -306,10 +354,13 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, if (ret < 0) goto dai_trigger_end; - ret = rsnd_platform_call(priv, dai, stop, ssi_id); + ret = rsnd_gen_path_exit(priv, rdai, io); if (ret < 0) goto dai_trigger_end; + ret = rsnd_platform_call(priv, dai, stop, ssi_id); + if (ret < 0) + goto dai_trigger_end; break; default: ret = -EINVAL; @@ -572,6 +623,10 @@ static int rsnd_probe(struct platform_device *pdev) /* * init each module */ + ret = rsnd_gen_probe(pdev, info, priv); + if (ret < 0) + return ret; + ret = rsnd_dai_probe(pdev, info, priv); if (ret < 0) return ret; @@ -615,6 +670,7 @@ static int rsnd_remove(struct platform_device *pdev) * remove each module */ rsnd_dai_remove(pdev, priv); + rsnd_gen_remove(pdev, priv); return 0; } diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c new file mode 100644 index 000000000000..ec67a796eca2 --- /dev/null +++ b/sound/soc/sh/rcar/gen.c @@ -0,0 +1,154 @@ +/* + * Renesas R-Car Gen1 SRU/SSI support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include "rsnd.h" + +struct rsnd_gen_ops { + int (*path_init)(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); + int (*path_exit)(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); +}; + +struct rsnd_gen_reg_map { + int index; /* -1 : not supported */ + u32 offset_id; /* offset of ssi0, ssi1, ssi2... */ + u32 offset_adr; /* offset of SSICR, SSISR, ... */ +}; + +struct rsnd_gen { + void __iomem *base[RSND_BASE_MAX]; + + struct rsnd_gen_reg_map reg_map[RSND_REG_MAX]; + struct rsnd_gen_ops *ops; +}; + +#define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) + +#define rsnd_is_gen1(s) ((s)->info->flags & RSND_GEN1) +#define rsnd_is_gen2(s) ((s)->info->flags & RSND_GEN2) + +/* + * Gen2 + * will be filled in the future + */ + +/* + * Gen1 + */ +static int rsnd_gen1_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + return 0; +} + +static void rsnd_gen1_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ +} + +/* + * Gen + */ +int rsnd_gen_path_init(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + return gen->ops->path_init(priv, rdai, io); +} + +int rsnd_gen_path_exit(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + return gen->ops->path_exit(priv, rdai, io); +} + +void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, + struct rsnd_mod *mod, + enum rsnd_reg reg) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + struct device *dev = rsnd_priv_to_dev(priv); + int index; + u32 offset_id, offset_adr; + + if (reg >= RSND_REG_MAX) { + dev_err(dev, "rsnd_reg reg error\n"); + return NULL; + } + + index = gen->reg_map[reg].index; + offset_id = gen->reg_map[reg].offset_id; + offset_adr = gen->reg_map[reg].offset_adr; + + if (index < 0) { + dev_err(dev, "unsupported reg access %d\n", reg); + return NULL; + } + + if (offset_id && mod) + offset_id *= rsnd_mod_id(mod); + + /* + * index/offset were set on gen1/gen2 + */ + + return gen->base[index] + offset_id + offset_adr; +} + +int rsnd_gen_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_gen *gen; + int i; + + gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL); + if (!gen) { + dev_err(dev, "GEN allocate failed\n"); + return -ENOMEM; + } + + priv->gen = gen; + + /* + * see + * rsnd_reg_get() + * rsnd_gen_probe() + */ + for (i = 0; i < RSND_REG_MAX; i++) + gen->reg_map[i].index = -1; + + /* + * init each module + */ + if (rsnd_is_gen1(priv)) + return rsnd_gen1_probe(pdev, info, priv); + + dev_err(dev, "unknown generation R-Car sound device\n"); + + return -ENODEV; +} + +void rsnd_gen_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ + if (rsnd_is_gen1(priv)) + rsnd_gen1_remove(pdev, priv); +} diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 65d3835cffbc..8cc36416da25 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -27,11 +27,35 @@ * This driver uses pseudo register in order to hide it. * see gen1/gen2 for detail */ +enum rsnd_reg { + RSND_REG_MAX, +}; + struct rsnd_priv; struct rsnd_mod; struct rsnd_dai; struct rsnd_dai_stream; +/* + * R-Car basic functions + */ +#define rsnd_mod_read(m, r) \ + rsnd_read(rsnd_mod_to_priv(m), m, RSND_REG_##r) +#define rsnd_mod_write(m, r, d) \ + rsnd_write(rsnd_mod_to_priv(m), m, RSND_REG_##r, d) +#define rsnd_mod_bset(m, r, s, d) \ + rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d) + +#define rsnd_priv_read(p, r) rsnd_read(p, NULL, RSND_REG_##r) +#define rsnd_priv_write(p, r, d) rsnd_write(p, NULL, RSND_REG_##r, d) +#define rsnd_priv_bset(p, r, s, d) rsnd_bset(p, NULL, RSND_REG_##r, s, d) + +u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); +void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data); +void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, + u32 mask, u32 data); + /* * R-Car sound mod */ @@ -116,6 +140,24 @@ int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io); void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int cnt); int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional); +/* + * R-Car Gen1/Gen2 + */ +int rsnd_gen_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv); +void rsnd_gen_remove(struct platform_device *pdev, + struct rsnd_priv *priv); +int rsnd_gen_path_init(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); +int rsnd_gen_path_exit(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io); +void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, + struct rsnd_mod *mod, + enum rsnd_reg reg); + /* * R-Car sound priv */ @@ -125,6 +167,11 @@ struct rsnd_priv { struct rcar_snd_info *info; spinlock_t lock; + /* + * below value will be filled on rsnd_gen_probe() + */ + void *gen; + /* * below value will be filled on rsnd_dai_probe() */ -- cgit v1.2.3-70-g09d2 From 07539c1de82cdc0ecbe72b413762b2e920407227 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 21 Jul 2013 21:36:35 -0700 Subject: ASoC: add Renesas R-Car SCU feature Renesas R-Car series sound circuit consists of SSI and its peripheral. But this peripheral circuit is different between R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2) (Actually, there are many difference in Generation1 chips) This patch adds SCU feature on this driver. But, it defines SCU style only, does nothing at this point. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 11 +++- sound/soc/sh/rcar/Makefile | 4 +- sound/soc/sh/rcar/core.c | 5 ++ sound/soc/sh/rcar/gen.c | 95 ++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 21 ++++++++ sound/soc/sh/rcar/scu.c | 125 +++++++++++++++++++++++++++++++++++++++++++++ 6 files changed, 258 insertions(+), 3 deletions(-) create mode 100644 sound/soc/sh/rcar/scu.c (limited to 'sound') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index 14942a827fe5..01f2e453dcbf 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -14,8 +14,15 @@ #include +#define RSND_GEN1_SRU 0 -#define RSND_BASE_MAX 0 +#define RSND_GEN2_SRU 0 + +#define RSND_BASE_MAX 1 + +struct rsnd_scu_platform_info { + u32 flags; +}; struct rsnd_dai_platform_info { int ssi_id_playback; @@ -34,6 +41,8 @@ struct rsnd_dai_platform_info { struct rcar_snd_info { u32 flags; + struct rsnd_scu_platform_info *scu_info; + int scu_info_nr; struct rsnd_dai_platform_info *dai_info; int dai_info_nr; int (*start)(int id); diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index b2d313b1eb94..112b2cfd793b 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,2 +1,2 @@ -snd-soc-rcar-objs := core.o gen.o -obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o +snd-soc-rcar-objs := core.o gen.o scu.o +obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o \ No newline at end of file diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index bb8959f93a7d..02d736bb4f54 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -631,6 +631,10 @@ static int rsnd_probe(struct platform_device *pdev) if (ret < 0) return ret; + ret = rsnd_scu_probe(pdev, info, priv); + if (ret < 0) + return ret; + /* * asoc register */ @@ -669,6 +673,7 @@ static int rsnd_remove(struct platform_device *pdev) /* * remove each module */ + rsnd_scu_remove(pdev, priv); rsnd_dai_remove(pdev, priv); rsnd_gen_remove(pdev, priv); diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ec67a796eca2..2934c0d731c8 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -45,10 +45,105 @@ struct rsnd_gen { /* * Gen1 */ +static int rsnd_gen1_path_init(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_dai_platform_info *info = rsnd_dai_get_platform_info(rdai); + struct rsnd_mod *mod; + int ret; + int id; + + /* + * Gen1 is created by SRU/SSI, and this SRU is base module of + * Gen2's SCU/SSIU/SSI. (Gen2 SCU/SSIU came from SRU) + * + * Easy image is.. + * Gen1 SRU = Gen2 SCU + SSIU + etc + * + * Gen2 SCU path is very flexible, but, Gen1 SRU (SCU parts) is + * using fixed path. + * + * Then, SSI id = SCU id here + */ + + if (rsnd_dai_is_play(rdai, io)) + id = info->ssi_id_playback; + else + id = info->ssi_id_capture; + + /* SCU */ + mod = rsnd_scu_mod_get(priv, id); + ret = rsnd_dai_connect(rdai, mod, io); + + return ret; +} + +static int rsnd_gen1_path_exit(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_mod *mod, *n; + int ret = 0; + + /* + * remove all mod from rdai + */ + for_each_rsnd_mod(mod, n, io) + ret |= rsnd_dai_disconnect(mod); + + return ret; +} + +static struct rsnd_gen_ops rsnd_gen1_ops = { + .path_init = rsnd_gen1_path_init, + .path_exit = rsnd_gen1_path_exit, +}; + +#define RSND_GEN1_REG_MAP(g, s, i, oi, oa) \ + do { \ + (g)->reg_map[RSND_REG_##i].index = RSND_GEN1_##s; \ + (g)->reg_map[RSND_REG_##i].offset_id = oi; \ + (g)->reg_map[RSND_REG_##i].offset_adr = oa; \ + } while (0) + +static void rsnd_gen1_reg_map_init(struct rsnd_gen *gen) +{ + RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE0, 0x0, 0xD0); + RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE1, 0x0, 0xD4); +} + static int rsnd_gen1_probe(struct platform_device *pdev, struct rcar_snd_info *info, struct rsnd_priv *priv) { + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + struct resource *sru_res; + + /* + * map address + */ + sru_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SRU); + if (!sru_res) { + dev_err(dev, "Not enough SRU/SSI/ADG platform resources.\n"); + return -ENODEV; + } + + gen->ops = &rsnd_gen1_ops; + + gen->base[RSND_GEN1_SRU] = devm_ioremap_resource(dev, sru_res); + if (!gen->base[RSND_GEN1_SRU]) { + dev_err(dev, "SRU/SSI/ADG ioremap failed\n"); + return -ENODEV; + } + + rsnd_gen1_reg_map_init(gen); + + dev_dbg(dev, "Gen1 device probed\n"); + dev_dbg(dev, "SRU : %08x => %p\n", sru_res->start, + gen->base[RSND_GEN1_SRU]); + return 0; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 8cc36416da25..95a391ff0627 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -28,6 +28,10 @@ * see gen1/gen2 for detail */ enum rsnd_reg { + /* SRU/SCU */ + RSND_REG_SSI_MODE0, + RSND_REG_SSI_MODE1, + RSND_REG_MAX, }; @@ -172,6 +176,12 @@ struct rsnd_priv { */ void *gen; + /* + * below value will be filled on rsnd_scu_probe() + */ + void *scu; + int scu_nr; + /* * below value will be filled on rsnd_dai_probe() */ @@ -184,4 +194,15 @@ struct rsnd_priv { #define rsnd_lock(priv, flags) spin_lock_irqsave(&priv->lock, flags) #define rsnd_unlock(priv, flags) spin_unlock_irqrestore(&priv->lock, flags) +/* + * R-Car SCU + */ +int rsnd_scu_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv); +void rsnd_scu_remove(struct platform_device *pdev, + struct rsnd_priv *priv); +struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id); +#define rsnd_scu_nr(priv) ((priv)->scu_nr) + #endif diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c new file mode 100644 index 000000000000..c12e65f240a1 --- /dev/null +++ b/sound/soc/sh/rcar/scu.c @@ -0,0 +1,125 @@ +/* + * Renesas R-Car SCU support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include "rsnd.h" + +struct rsnd_scu { + struct rsnd_scu_platform_info *info; /* rcar_snd.h */ + struct rsnd_mod mod; +}; + +#define rsnd_mod_to_scu(_mod) \ + container_of((_mod), struct rsnd_scu, mod) + +#define for_each_rsnd_scu(pos, priv, i) \ + for ((i) = 0; \ + ((i) < rsnd_scu_nr(priv)) && \ + ((pos) = (struct rsnd_scu *)(priv)->scu + i); \ + i++) + +static int rsnd_scu_init(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; +} + +static int rsnd_scu_quit(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_dbg(dev, "%s.%d quit\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; +} + +static int rsnd_scu_start(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; +} + +static int rsnd_scu_stop(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_dbg(dev, "%s.%d stop\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; +} + +static struct rsnd_mod_ops rsnd_scu_ops = { + .name = "scu", + .init = rsnd_scu_init, + .quit = rsnd_scu_quit, + .start = rsnd_scu_start, + .stop = rsnd_scu_stop, +}; + +struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id) +{ + BUG_ON(id < 0 || id >= rsnd_scu_nr(priv)); + + return &((struct rsnd_scu *)(priv->scu) + id)->mod; +} + +int rsnd_scu_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_scu *scu; + int i, nr; + + /* + * init SCU + */ + nr = info->scu_info_nr; + scu = devm_kzalloc(dev, sizeof(*scu) * nr, GFP_KERNEL); + if (!scu) { + dev_err(dev, "SCU allocate failed\n"); + return -ENOMEM; + } + + priv->scu_nr = nr; + priv->scu = scu; + + for_each_rsnd_scu(scu, priv, i) { + rsnd_mod_init(priv, &scu->mod, + &rsnd_scu_ops, i); + scu->info = &info->scu_info[i]; + } + + dev_dbg(dev, "scu probed\n"); + + return 0; +} + +void rsnd_scu_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ +} -- cgit v1.2.3-70-g09d2 From dfc9403b7c1f566bb099a12c58aee20589e390f1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 21 Jul 2013 21:36:46 -0700 Subject: ASoC: add Renesas R-Car ADG feature Renesas R-Car series sound circuit consists of SSI and its peripheral. But this peripheral circuit is different between R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2) (Actually, there are many difference in Generation1 chips) This patch adds ADG feature which controls sound clock Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 4 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 234 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/core.c | 5 + sound/soc/sh/rcar/gen.c | 20 +++- sound/soc/sh/rcar/rsnd.h | 27 ++++++ 6 files changed, 288 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sh/rcar/adg.c (limited to 'sound') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index 01f2e453dcbf..6babd6f7b537 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -15,10 +15,12 @@ #include #define RSND_GEN1_SRU 0 +#define RSND_GEN1_ADG 1 #define RSND_GEN2_SRU 0 +#define RSND_GEN2_ADG 1 -#define RSND_BASE_MAX 1 +#define RSND_BASE_MAX 2 struct rsnd_scu_platform_info { u32 flags; diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index 112b2cfd793b..c11280cffcfe 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,2 +1,2 @@ -snd-soc-rcar-objs := core.o gen.o scu.o +snd-soc-rcar-objs := core.o gen.o scu.o adg.o obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o \ No newline at end of file diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c new file mode 100644 index 000000000000..d80deb7ccf13 --- /dev/null +++ b/sound/soc/sh/rcar/adg.c @@ -0,0 +1,234 @@ +/* + * Helper routines for R-Car sound ADG. + * + * Copyright (C) 2013 Kuninori Morimoto + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + */ +#include +#include +#include "rsnd.h" + +#define CLKA 0 +#define CLKB 1 +#define CLKC 2 +#define CLKI 3 +#define CLKMAX 4 + +struct rsnd_adg { + struct clk *clk[CLKMAX]; + + int rate_of_441khz_div_6; + int rate_of_48khz_div_6; +}; + +#define for_each_rsnd_clk(pos, adg, i) \ + for (i = 0, (pos) = adg->clk[i]; \ + i < CLKMAX; \ + i++, (pos) = adg->clk[i]) +#define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) + +static enum rsnd_reg rsnd_adg_ssi_reg_get(int id) +{ + enum rsnd_reg reg; + + /* + * SSI 8 is not connected to ADG. + * it works with SSI 7 + */ + if (id == 8) + return RSND_REG_MAX; + + if (0 <= id && id <= 3) + reg = RSND_REG_AUDIO_CLK_SEL0; + else if (4 <= id && id <= 7) + reg = RSND_REG_AUDIO_CLK_SEL1; + else + reg = RSND_REG_AUDIO_CLK_SEL2; + + return reg; +} + +int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + enum rsnd_reg reg; + int id; + + /* + * "mod" = "ssi" here. + * we can get "ssi id" from mod + */ + id = rsnd_mod_id(mod); + reg = rsnd_adg_ssi_reg_get(id); + + rsnd_write(priv, mod, reg, 0); + + return 0; +} + +int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_adg *adg = rsnd_priv_to_adg(priv); + struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; + enum rsnd_reg reg; + int id, shift, i; + u32 data; + int sel_table[] = { + [CLKA] = 0x1, + [CLKB] = 0x2, + [CLKC] = 0x3, + [CLKI] = 0x0, + }; + + dev_dbg(dev, "request clock = %d\n", rate); + + /* + * find suitable clock from + * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI. + */ + data = 0; + for_each_rsnd_clk(clk, adg, i) { + if (rate == clk_get_rate(clk)) { + data = sel_table[i]; + goto found_clock; + } + } + + /* + * find 1/6 clock from BRGA/BRGB + */ + if (rate == adg->rate_of_441khz_div_6) { + data = 0x10; + goto found_clock; + } + + if (rate == adg->rate_of_48khz_div_6) { + data = 0x20; + goto found_clock; + } + + return -EIO; + +found_clock: + + /* + * This "mod" = "ssi" here. + * we can get "ssi id" from mod + */ + id = rsnd_mod_id(mod); + reg = rsnd_adg_ssi_reg_get(id); + + dev_dbg(dev, "ADG: ssi%d selects clk%d = %d", id, i, rate); + + /* + * Enable SSIx clock + */ + shift = (id % 4) * 8; + + rsnd_bset(priv, mod, reg, + 0xFF << shift, + data << shift); + + return 0; +} + +static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) +{ + struct clk *clk; + unsigned long rate; + u32 ckr; + int i; + int brg_table[] = { + [CLKA] = 0x0, + [CLKB] = 0x1, + [CLKC] = 0x4, + [CLKI] = 0x2, + }; + + /* + * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC + * have 44.1kHz or 48kHz base clocks for now. + * + * SSI itself can divide parent clock by 1/1 - 1/16 + * So, BRGA outputs 44.1kHz base parent clock 1/32, + * and, BRGB outputs 48.0kHz base parent clock 1/32 here. + * see + * rsnd_adg_ssi_clk_try_start() + */ + ckr = 0; + adg->rate_of_441khz_div_6 = 0; + adg->rate_of_48khz_div_6 = 0; + for_each_rsnd_clk(clk, adg, i) { + rate = clk_get_rate(clk); + + if (0 == rate) /* not used */ + continue; + + /* RBGA */ + if (!adg->rate_of_441khz_div_6 && (0 == rate % 44100)) { + adg->rate_of_441khz_div_6 = rate / 6; + ckr |= brg_table[i] << 20; + } + + /* RBGB */ + if (!adg->rate_of_48khz_div_6 && (0 == rate % 48000)) { + adg->rate_of_48khz_div_6 = rate / 6; + ckr |= brg_table[i] << 16; + } + } + + rsnd_priv_bset(priv, SSICKR, 0x00FF0000, ckr); + rsnd_priv_write(priv, BRRA, 0x00000002); /* 1/6 */ + rsnd_priv_write(priv, BRRB, 0x00000002); /* 1/6 */ +} + +int rsnd_adg_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct rsnd_adg *adg; + struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; + int i; + + adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); + if (!adg) { + dev_err(dev, "ADG allocate failed\n"); + return -ENOMEM; + } + + adg->clk[CLKA] = clk_get(NULL, "audio_clk_a"); + adg->clk[CLKB] = clk_get(NULL, "audio_clk_b"); + adg->clk[CLKC] = clk_get(NULL, "audio_clk_c"); + adg->clk[CLKI] = clk_get(NULL, "audio_clk_internal"); + for_each_rsnd_clk(clk, adg, i) { + if (IS_ERR(clk)) { + dev_err(dev, "Audio clock failed\n"); + return -EIO; + } + } + + rsnd_adg_ssi_clk_init(priv, adg); + + priv->adg = adg; + + dev_dbg(dev, "adg probed\n"); + + return 0; +} + +void rsnd_adg_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ + struct rsnd_adg *adg = priv->adg; + struct clk *clk; + int i; + + for_each_rsnd_clk(clk, adg, i) + clk_put(clk); +} diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 02d736bb4f54..e588d8a8ae40 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -635,6 +635,10 @@ static int rsnd_probe(struct platform_device *pdev) if (ret < 0) return ret; + ret = rsnd_adg_probe(pdev, info, priv); + if (ret < 0) + return ret; + /* * asoc register */ @@ -673,6 +677,7 @@ static int rsnd_remove(struct platform_device *pdev) /* * remove each module */ + rsnd_adg_remove(pdev, priv); rsnd_scu_remove(pdev, priv); rsnd_dai_remove(pdev, priv); rsnd_gen_remove(pdev, priv); diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 2934c0d731c8..ed21a136354f 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -111,6 +111,15 @@ static void rsnd_gen1_reg_map_init(struct rsnd_gen *gen) { RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE0, 0x0, 0xD0); RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE1, 0x0, 0xD4); + + RSND_GEN1_REG_MAP(gen, ADG, BRRA, 0x0, 0x00); + RSND_GEN1_REG_MAP(gen, ADG, BRRB, 0x0, 0x04); + RSND_GEN1_REG_MAP(gen, ADG, SSICKR, 0x0, 0x08); + RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL0, 0x0, 0x0c); + RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL1, 0x0, 0x10); + RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL3, 0x0, 0x18); + RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL4, 0x0, 0x1c); + RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL5, 0x0, 0x20); } static int rsnd_gen1_probe(struct platform_device *pdev, @@ -120,12 +129,15 @@ static int rsnd_gen1_probe(struct platform_device *pdev, struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); struct resource *sru_res; + struct resource *adg_res; /* * map address */ sru_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SRU); - if (!sru_res) { + adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_ADG); + if (!sru_res || + !adg_res) { dev_err(dev, "Not enough SRU/SSI/ADG platform resources.\n"); return -ENODEV; } @@ -133,7 +145,9 @@ static int rsnd_gen1_probe(struct platform_device *pdev, gen->ops = &rsnd_gen1_ops; gen->base[RSND_GEN1_SRU] = devm_ioremap_resource(dev, sru_res); - if (!gen->base[RSND_GEN1_SRU]) { + gen->base[RSND_GEN1_ADG] = devm_ioremap_resource(dev, adg_res); + if (!gen->base[RSND_GEN1_SRU] || + !gen->base[RSND_GEN1_ADG]) { dev_err(dev, "SRU/SSI/ADG ioremap failed\n"); return -ENODEV; } @@ -143,6 +157,8 @@ static int rsnd_gen1_probe(struct platform_device *pdev, dev_dbg(dev, "Gen1 device probed\n"); dev_dbg(dev, "SRU : %08x => %p\n", sru_res->start, gen->base[RSND_GEN1_SRU]); + dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + gen->base[RSND_GEN1_ADG]); return 0; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 95a391ff0627..344fd59cb7fd 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -32,6 +32,17 @@ enum rsnd_reg { RSND_REG_SSI_MODE0, RSND_REG_SSI_MODE1, + /* ADG */ + RSND_REG_BRRA, + RSND_REG_BRRB, + RSND_REG_SSICKR, + RSND_REG_AUDIO_CLK_SEL0, + RSND_REG_AUDIO_CLK_SEL1, + RSND_REG_AUDIO_CLK_SEL2, + RSND_REG_AUDIO_CLK_SEL3, + RSND_REG_AUDIO_CLK_SEL4, + RSND_REG_AUDIO_CLK_SEL5, + RSND_REG_MAX, }; @@ -162,6 +173,17 @@ void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); +/* + * R-Car ADG + */ +int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod); +int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate); +int rsnd_adg_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv); +void rsnd_adg_remove(struct platform_device *pdev, + struct rsnd_priv *priv); + /* * R-Car sound priv */ @@ -182,6 +204,11 @@ struct rsnd_priv { void *scu; int scu_nr; + /* + * below value will be filled on rsnd_adg_probe() + */ + void *adg; + /* * below value will be filled on rsnd_dai_probe() */ -- cgit v1.2.3-70-g09d2 From ae5c322303fff50b93d60e34c6563f1264a5941b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 21 Jul 2013 21:36:57 -0700 Subject: ASoC: add Renesas R-Car SSI feature Renesas R-Car series sound circuit consists of SSI and its peripheral. But this peripheral circuit is different between R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2) (Actually, there are many difference in Generation1 chips) As 1st protype, this patch adds SSI feature on this driver. But, it is PIO sound playback support only at this point. The DMA transfer, and capture feature will be supported in the future Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 23 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 5 + sound/soc/sh/rcar/gen.c | 24 +- sound/soc/sh/rcar/rsnd.h | 23 ++ sound/soc/sh/rcar/ssi.c | 588 +++++++++++++++++++++++++++++++++++++++++++++ 6 files changed, 661 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sh/rcar/ssi.c (limited to 'sound') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index 6babd6f7b537..99d8dd029906 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -16,11 +16,30 @@ #define RSND_GEN1_SRU 0 #define RSND_GEN1_ADG 1 +#define RSND_GEN1_SSI 2 #define RSND_GEN2_SRU 0 #define RSND_GEN2_ADG 1 +#define RSND_GEN2_SSIU 2 +#define RSND_GEN2_SSI 3 -#define RSND_BASE_MAX 2 +#define RSND_BASE_MAX 4 + +/* + * flags + * + * 0xA0000000 + * + * A : clock sharing settings + */ +#define RSND_SSI_CLK_PIN_SHARE (1 << 31) +#define RSND_SSI_CLK_FROM_ADG (1 << 30) /* clock parent is master */ +#define RSND_SSI_SYNC (1 << 29) /* SSI34_sync etc */ + +struct rsnd_ssi_platform_info { + int pio_irq; + u32 flags; +}; struct rsnd_scu_platform_info { u32 flags; @@ -43,6 +62,8 @@ struct rsnd_dai_platform_info { struct rcar_snd_info { u32 flags; + struct rsnd_ssi_platform_info *ssi_info; + int ssi_info_nr; struct rsnd_scu_platform_info *scu_info; int scu_info_nr; struct rsnd_dai_platform_info *dai_info; diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index c11280cffcfe..0ff492df7929 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,2 +1,2 @@ -snd-soc-rcar-objs := core.o gen.o scu.o adg.o +snd-soc-rcar-objs := core.o gen.o scu.o adg.o ssi.o obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o \ No newline at end of file diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e588d8a8ae40..9a5469d3f352 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -639,6 +639,10 @@ static int rsnd_probe(struct platform_device *pdev) if (ret < 0) return ret; + ret = rsnd_ssi_probe(pdev, info, priv); + if (ret < 0) + return ret; + /* * asoc register */ @@ -677,6 +681,7 @@ static int rsnd_remove(struct platform_device *pdev) /* * remove each module */ + rsnd_ssi_remove(pdev, priv); rsnd_adg_remove(pdev, priv); rsnd_scu_remove(pdev, priv); rsnd_dai_remove(pdev, priv); diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ed21a136354f..5e4ae0da4352 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -72,6 +72,12 @@ static int rsnd_gen1_path_init(struct rsnd_priv *priv, else id = info->ssi_id_capture; + /* SSI */ + mod = rsnd_ssi_mod_get(priv, id); + ret = rsnd_dai_connect(rdai, mod, io); + if (ret < 0) + return ret; + /* SCU */ mod = rsnd_scu_mod_get(priv, id); ret = rsnd_dai_connect(rdai, mod, io); @@ -120,6 +126,12 @@ static void rsnd_gen1_reg_map_init(struct rsnd_gen *gen) RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL3, 0x0, 0x18); RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL4, 0x0, 0x1c); RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL5, 0x0, 0x20); + + RSND_GEN1_REG_MAP(gen, SSI, SSICR, 0x40, 0x00); + RSND_GEN1_REG_MAP(gen, SSI, SSISR, 0x40, 0x04); + RSND_GEN1_REG_MAP(gen, SSI, SSITDR, 0x40, 0x08); + RSND_GEN1_REG_MAP(gen, SSI, SSIRDR, 0x40, 0x0c); + RSND_GEN1_REG_MAP(gen, SSI, SSIWSR, 0x40, 0x20); } static int rsnd_gen1_probe(struct platform_device *pdev, @@ -130,14 +142,17 @@ static int rsnd_gen1_probe(struct platform_device *pdev, struct rsnd_gen *gen = rsnd_priv_to_gen(priv); struct resource *sru_res; struct resource *adg_res; + struct resource *ssi_res; /* * map address */ sru_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SRU); adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_ADG); + ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SSI); if (!sru_res || - !adg_res) { + !adg_res || + !ssi_res) { dev_err(dev, "Not enough SRU/SSI/ADG platform resources.\n"); return -ENODEV; } @@ -146,8 +161,10 @@ static int rsnd_gen1_probe(struct platform_device *pdev, gen->base[RSND_GEN1_SRU] = devm_ioremap_resource(dev, sru_res); gen->base[RSND_GEN1_ADG] = devm_ioremap_resource(dev, adg_res); + gen->base[RSND_GEN1_SSI] = devm_ioremap_resource(dev, ssi_res); if (!gen->base[RSND_GEN1_SRU] || - !gen->base[RSND_GEN1_ADG]) { + !gen->base[RSND_GEN1_ADG] || + !gen->base[RSND_GEN1_SSI]) { dev_err(dev, "SRU/SSI/ADG ioremap failed\n"); return -ENODEV; } @@ -159,8 +176,11 @@ static int rsnd_gen1_probe(struct platform_device *pdev, gen->base[RSND_GEN1_SRU]); dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, gen->base[RSND_GEN1_ADG]); + dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + gen->base[RSND_GEN1_SSI]); return 0; + } static void rsnd_gen1_remove(struct platform_device *pdev, diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 344fd59cb7fd..0e7727cc41db 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -43,6 +43,13 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL4, RSND_REG_AUDIO_CLK_SEL5, + /* SSI */ + RSND_REG_SSICR, + RSND_REG_SSISR, + RSND_REG_SSITDR, + RSND_REG_SSIRDR, + RSND_REG_SSIWSR, + RSND_REG_MAX, }; @@ -151,6 +158,7 @@ int rsnd_dai_connect(struct rsnd_dai *rdai, struct rsnd_mod *mod, struct rsnd_dai_stream *io); int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io); #define rsnd_dai_get_platform_info(rdai) ((rdai)->info) +#define rsnd_io_to_runtime(io) ((io)->substream->runtime) void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int cnt); int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional); @@ -209,6 +217,11 @@ struct rsnd_priv { */ void *adg; + /* + * below value will be filled on rsnd_ssi_probe() + */ + void *ssiu; + /* * below value will be filled on rsnd_dai_probe() */ @@ -232,4 +245,14 @@ void rsnd_scu_remove(struct platform_device *pdev, struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id); #define rsnd_scu_nr(priv) ((priv)->scu_nr) +/* + * R-Car SSI + */ +int rsnd_ssi_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv); +void rsnd_ssi_remove(struct platform_device *pdev, + struct rsnd_priv *priv); +struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); + #endif diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c new file mode 100644 index 000000000000..061ac7e88309 --- /dev/null +++ b/sound/soc/sh/rcar/ssi.c @@ -0,0 +1,588 @@ +/* + * Renesas R-Car SSIU/SSI support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto + * + * Based on fsi.c + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include +#include "rsnd.h" +#define RSND_SSI_NAME_SIZE 16 + +/* + * SSICR + */ +#define FORCE (1 << 31) /* Fixed */ +#define UIEN (1 << 27) /* Underflow Interrupt Enable */ +#define OIEN (1 << 26) /* Overflow Interrupt Enable */ +#define IIEN (1 << 25) /* Idle Mode Interrupt Enable */ +#define DIEN (1 << 24) /* Data Interrupt Enable */ + +#define DWL_8 (0 << 19) /* Data Word Length */ +#define DWL_16 (1 << 19) /* Data Word Length */ +#define DWL_18 (2 << 19) /* Data Word Length */ +#define DWL_20 (3 << 19) /* Data Word Length */ +#define DWL_22 (4 << 19) /* Data Word Length */ +#define DWL_24 (5 << 19) /* Data Word Length */ +#define DWL_32 (6 << 19) /* Data Word Length */ + +#define SWL_32 (3 << 16) /* R/W System Word Length */ +#define SCKD (1 << 15) /* Serial Bit Clock Direction */ +#define SWSD (1 << 14) /* Serial WS Direction */ +#define SCKP (1 << 13) /* Serial Bit Clock Polarity */ +#define SWSP (1 << 12) /* Serial WS Polarity */ +#define SDTA (1 << 10) /* Serial Data Alignment */ +#define DEL (1 << 8) /* Serial Data Delay */ +#define CKDV(v) (v << 4) /* Serial Clock Division Ratio */ +#define TRMD (1 << 1) /* Transmit/Receive Mode Select */ +#define EN (1 << 0) /* SSI Module Enable */ + +/* + * SSISR + */ +#define UIRQ (1 << 27) /* Underflow Error Interrupt Status */ +#define OIRQ (1 << 26) /* Overflow Error Interrupt Status */ +#define IIRQ (1 << 25) /* Idle Mode Interrupt Status */ +#define DIRQ (1 << 24) /* Data Interrupt Status Flag */ + +struct rsnd_ssi { + struct clk *clk; + struct rsnd_ssi_platform_info *info; /* rcar_snd.h */ + struct rsnd_ssi *parent; + struct rsnd_mod mod; + + struct rsnd_dai *rdai; + struct rsnd_dai_stream *io; + u32 cr_own; + u32 cr_clk; + u32 cr_etc; + int err; + unsigned int usrcnt; + unsigned int rate; +}; + +struct rsnd_ssiu { + u32 ssi_mode0; + u32 ssi_mode1; + + int ssi_nr; + struct rsnd_ssi *ssi; +}; + +#define for_each_rsnd_ssi(pos, priv, i) \ + for (i = 0; \ + (i < rsnd_ssi_nr(priv)) && \ + ((pos) = ((struct rsnd_ssiu *)((priv)->ssiu))->ssi + i); \ + i++) + +#define rsnd_ssi_nr(priv) (((struct rsnd_ssiu *)((priv)->ssiu))->ssi_nr) +#define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod) +#define rsnd_ssi_is_pio(ssi) ((ssi)->info->pio_irq > 0) +#define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent) +#define rsnd_rdai_is_clk_master(rdai) ((rdai)->clk_master) +#define rsnd_ssi_mode_flags(p) ((p)->info->flags) +#define rsnd_ssi_to_ssiu(ssi)\ + (((struct rsnd_ssiu *)((ssi) - rsnd_mod_id(&(ssi)->mod))) - 1) + +static void rsnd_ssi_mode_init(struct rsnd_priv *priv, + struct rsnd_ssiu *ssiu) +{ + struct rsnd_ssi *ssi; + u32 flags; + u32 val; + int i; + + /* + * SSI_MODE0 + */ + ssiu->ssi_mode0 = 0; + for_each_rsnd_ssi(ssi, priv, i) + ssiu->ssi_mode0 |= (1 << i); + + /* + * SSI_MODE1 + */ +#define ssi_parent_set(p, sync, adg, ext) \ + do { \ + ssi->parent = ssiu->ssi + p; \ + if (flags & RSND_SSI_CLK_FROM_ADG) \ + val = adg; \ + else \ + val = ext; \ + if (flags & RSND_SSI_SYNC) \ + val |= sync; \ + } while (0) + + ssiu->ssi_mode1 = 0; + for_each_rsnd_ssi(ssi, priv, i) { + flags = rsnd_ssi_mode_flags(ssi); + + if (!(flags & RSND_SSI_CLK_PIN_SHARE)) + continue; + + val = 0; + switch (i) { + case 1: + ssi_parent_set(0, (1 << 4), (0x2 << 0), (0x1 << 0)); + break; + case 2: + ssi_parent_set(0, (1 << 4), (0x2 << 2), (0x1 << 2)); + break; + case 4: + ssi_parent_set(3, (1 << 20), (0x2 << 16), (0x1 << 16)); + break; + case 8: + ssi_parent_set(7, 0, 0, 0); + break; + } + + ssiu->ssi_mode1 |= val; + } +} + +static void rsnd_ssi_mode_set(struct rsnd_ssi *ssi) +{ + struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi); + + rsnd_mod_write(&ssi->mod, SSI_MODE0, ssiu->ssi_mode0); + rsnd_mod_write(&ssi->mod, SSI_MODE1, ssiu->ssi_mode1); +} + +static void rsnd_ssi_status_check(struct rsnd_mod *mod, + u32 bit) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + u32 status; + int i; + + for (i = 0; i < 1024; i++) { + status = rsnd_mod_read(mod, SSISR); + if (status & bit) + return; + + udelay(50); + } + + dev_warn(dev, "status check failed\n"); +} + +static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, + unsigned int rate) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod); + struct device *dev = rsnd_priv_to_dev(priv); + int i, j, ret; + int adg_clk_div_table[] = { + 1, 6, /* see adg.c */ + }; + int ssi_clk_mul_table[] = { + 1, 2, 4, 8, 16, 6, 12, + }; + unsigned int main_rate; + + /* + * Find best clock, and try to start ADG + */ + for (i = 0; i < ARRAY_SIZE(adg_clk_div_table); i++) { + for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) { + + /* + * this driver is assuming that + * system word is 64fs (= 2 x 32bit) + * see rsnd_ssi_start() + */ + main_rate = rate / adg_clk_div_table[i] + * 32 * 2 * ssi_clk_mul_table[j]; + + ret = rsnd_adg_ssi_clk_try_start(&ssi->mod, main_rate); + if (0 == ret) { + ssi->rate = rate; + ssi->cr_clk = FORCE | SWL_32 | + SCKD | SWSD | CKDV(j); + + dev_dbg(dev, "ssi%d outputs %u Hz\n", + rsnd_mod_id(&ssi->mod), rate); + + return 0; + } + } + } + + dev_err(dev, "unsupported clock rate\n"); + return -EIO; +} + +static void rsnd_ssi_master_clk_stop(struct rsnd_ssi *ssi) +{ + ssi->rate = 0; + ssi->cr_clk = 0; + rsnd_adg_ssi_clk_stop(&ssi->mod); +} + +static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod); + struct device *dev = rsnd_priv_to_dev(priv); + u32 cr; + + if (0 == ssi->usrcnt) { + clk_enable(ssi->clk); + + if (rsnd_rdai_is_clk_master(rdai)) { + struct snd_pcm_runtime *runtime; + + runtime = rsnd_io_to_runtime(io); + + if (rsnd_ssi_clk_from_parent(ssi)) + rsnd_ssi_hw_start(ssi->parent, rdai, io); + else + rsnd_ssi_master_clk_start(ssi, runtime->rate); + } + } + + cr = ssi->cr_own | + ssi->cr_clk | + ssi->cr_etc | + EN; + + rsnd_mod_write(&ssi->mod, SSICR, cr); + + ssi->usrcnt++; + + dev_dbg(dev, "ssi%d hw started\n", rsnd_mod_id(&ssi->mod)); +} + +static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, + struct rsnd_dai *rdai) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod); + struct device *dev = rsnd_priv_to_dev(priv); + u32 cr; + + if (0 == ssi->usrcnt) /* stop might be called without start */ + return; + + ssi->usrcnt--; + + if (0 == ssi->usrcnt) { + /* + * disable all IRQ, + * and, wait all data was sent + */ + cr = ssi->cr_own | + ssi->cr_clk; + + rsnd_mod_write(&ssi->mod, SSICR, cr | EN); + rsnd_ssi_status_check(&ssi->mod, DIRQ); + + /* + * disable SSI, + * and, wait idle state + */ + rsnd_mod_write(&ssi->mod, SSICR, cr); /* disabled all */ + rsnd_ssi_status_check(&ssi->mod, IIRQ); + + if (rsnd_rdai_is_clk_master(rdai)) { + if (rsnd_ssi_clk_from_parent(ssi)) + rsnd_ssi_hw_stop(ssi->parent, rdai); + else + rsnd_ssi_master_clk_stop(ssi); + } + + clk_disable(ssi->clk); + } + + dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod)); +} + +/* + * SSI mod common functions + */ +static int rsnd_ssi_init(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + u32 cr; + + cr = FORCE; + + /* + * always use 32bit system word for easy clock calculation. + * see also rsnd_ssi_master_clk_enable() + */ + cr |= SWL_32; + + /* + * init clock settings for SSICR + */ + switch (runtime->sample_bits) { + case 16: + cr |= DWL_16; + break; + case 32: + cr |= DWL_24; + break; + default: + return -EIO; + } + + if (rdai->bit_clk_inv) + cr |= SCKP; + if (rdai->frm_clk_inv) + cr |= SWSP; + if (rdai->data_alignment) + cr |= SDTA; + if (rdai->sys_delay) + cr |= DEL; + if (rsnd_dai_is_play(rdai, io)) + cr |= TRMD; + + /* + * set ssi parameter + */ + ssi->rdai = rdai; + ssi->io = io; + ssi->cr_own = cr; + ssi->err = -1; /* ignore 1st error */ + + rsnd_ssi_mode_set(ssi); + + dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; +} + +static int rsnd_ssi_quit(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_dbg(dev, "%s.%d quit\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + if (ssi->err > 0) + dev_warn(dev, "ssi under/over flow err = %d\n", ssi->err); + + ssi->rdai = NULL; + ssi->io = NULL; + ssi->cr_own = 0; + ssi->err = 0; + + return 0; +} + +static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status) +{ + /* under/over flow error */ + if (status & (UIRQ | OIRQ)) { + ssi->err++; + + /* clear error status */ + rsnd_mod_write(&ssi->mod, SSISR, 0); + } +} + +/* + * SSI PIO + */ +static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data) +{ + struct rsnd_ssi *ssi = data; + struct rsnd_dai_stream *io = ssi->io; + u32 status = rsnd_mod_read(&ssi->mod, SSISR); + irqreturn_t ret = IRQ_NONE; + + if (io && (status & DIRQ)) { + struct rsnd_dai *rdai = ssi->rdai; + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + u32 *buf = (u32 *)(runtime->dma_area + + rsnd_dai_pointer_offset(io, 0)); + + rsnd_ssi_record_error(ssi, status); + + /* + * 8/16/32 data can be assesse to TDR/RDR register + * directly as 32bit data + * see rsnd_ssi_init() + */ + if (rsnd_dai_is_play(rdai, io)) + rsnd_mod_write(&ssi->mod, SSITDR, *buf); + else + *buf = rsnd_mod_read(&ssi->mod, SSIRDR); + + rsnd_dai_pointer_update(io, sizeof(*buf)); + + ret = IRQ_HANDLED; + } + + return ret; +} + +static int rsnd_ssi_pio_start(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + /* enable PIO IRQ */ + ssi->cr_etc = UIEN | OIEN | DIEN; + + rsnd_ssi_hw_start(ssi, rdai, io); + + dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; +} + +static int rsnd_ssi_pio_stop(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + + dev_dbg(dev, "%s.%d stop\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + ssi->cr_etc = 0; + + rsnd_ssi_hw_stop(ssi, rdai); + + return 0; +} + +static struct rsnd_mod_ops rsnd_ssi_pio_ops = { + .name = "ssi (pio)", + .init = rsnd_ssi_init, + .quit = rsnd_ssi_quit, + .start = rsnd_ssi_pio_start, + .stop = rsnd_ssi_pio_stop, +}; + +/* + * Non SSI + */ +static int rsnd_ssi_non(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_dbg(dev, "%s\n", __func__); + + return 0; +} + +static struct rsnd_mod_ops rsnd_ssi_non_ops = { + .name = "ssi (non)", + .init = rsnd_ssi_non, + .quit = rsnd_ssi_non, + .start = rsnd_ssi_non, + .stop = rsnd_ssi_non, +}; + +/* + * ssi mod function + */ +struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id) +{ + BUG_ON(id < 0 || id >= rsnd_ssi_nr(priv)); + + return &(((struct rsnd_ssiu *)(priv->ssiu))->ssi + id)->mod; +} + +int rsnd_ssi_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct rsnd_ssi_platform_info *pinfo; + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_mod_ops *ops; + struct clk *clk; + struct rsnd_ssiu *ssiu; + struct rsnd_ssi *ssi; + char name[RSND_SSI_NAME_SIZE]; + int i, nr, ret; + + /* + * init SSI + */ + nr = info->ssi_info_nr; + ssiu = devm_kzalloc(dev, sizeof(*ssiu) + (sizeof(*ssi) * nr), + GFP_KERNEL); + if (!ssiu) { + dev_err(dev, "SSI allocate failed\n"); + return -ENOMEM; + } + + priv->ssiu = ssiu; + ssiu->ssi = (struct rsnd_ssi *)(ssiu + 1); + ssiu->ssi_nr = nr; + + for_each_rsnd_ssi(ssi, priv, i) { + pinfo = &info->ssi_info[i]; + + snprintf(name, RSND_SSI_NAME_SIZE, "ssi.%d", i); + + clk = clk_get(dev, name); + if (IS_ERR(clk)) + return PTR_ERR(clk); + + ssi->info = pinfo; + ssi->clk = clk; + + ops = &rsnd_ssi_non_ops; + + /* + * SSI PIO case + */ + if (rsnd_ssi_is_pio(ssi)) { + ret = devm_request_irq(dev, pinfo->pio_irq, + &rsnd_ssi_pio_interrupt, + IRQF_SHARED, + dev_name(dev), ssi); + if (ret) { + dev_err(dev, "SSI request interrupt failed\n"); + return ret; + } + + ops = &rsnd_ssi_pio_ops; + } + + rsnd_mod_init(priv, &ssi->mod, ops, i); + } + + rsnd_ssi_mode_init(priv, ssiu); + + dev_dbg(dev, "ssi probed\n"); + + return 0; +} + +void rsnd_ssi_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ + struct rsnd_ssi *ssi; + int i; + + for_each_rsnd_ssi(ssi, priv, i) + clk_put(ssi->clk); +} -- cgit v1.2.3-70-g09d2 From a8d30608eaed6cc759b8e2e8a8bbbb42591f797f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 29 Jul 2013 15:10:22 +0530 Subject: ALSA: compress: fix the return value for SNDRV_COMPRESS_VERSION the return value of SNDRV_COMPRESS_VERSION always return default -ENOTTY as the return value was never updated for this call assign return value from put_user() Reported-by: Haynes CC: stable@vger.kernel.org Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892d7299..98969541cbcc 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -743,7 +743,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - put_user(SNDRV_COMPRESS_VERSION, + retval = put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): -- cgit v1.2.3-70-g09d2 From 4fefd69853a4e83040ddaa98d3b6e5e12cc4f97a Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Mon, 29 Jul 2013 13:51:58 +0100 Subject: ASoC: core: Add snd_soc_card_get_kcontrol() This is useful for drivers who want to grab a pointer to snd_kcontrol outside of the kcontrol callbacks. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 16 ++++++++++++++++ 2 files changed, 18 insertions(+) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 6eabee7ec15a..b33d1de46396 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -475,6 +475,8 @@ int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, void *data, const char *long_name, const char *prefix); +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name); int snd_soc_add_codec_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_add_platform_controls(struct snd_soc_platform *platform, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0ec070cf7231..cef714effc1e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2299,6 +2299,22 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev, return 0; } +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name) +{ + struct snd_card *card = soc_card->snd_card; + struct snd_kcontrol *kctl; + + if (unlikely(!name)) + return NULL; + + list_for_each_entry(kctl, &card->controls, list) + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) + return kctl; + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); + /** * snd_soc_add_codec_controls - add an array of controls to a codec. * Convenience function to add a list of controls. Many codecs were -- cgit v1.2.3-70-g09d2 From 81ad93ecfda64cb37129d29adb384affd0d0fa5b Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Mon, 29 Jul 2013 13:51:59 +0100 Subject: ASoC: wm_adsp: Simplify kcontrol handling Get rid off the wm_coeff struct and the wm_coeff_add_kcontrol() function. We are now using the snd_soc_card_kcontrol() function to get the kcontrol pointers. No need to call into ALSA code to register the kcontrols. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 103 ++++++++++++++------------------------------- sound/soc/codecs/wm_adsp.h | 2 +- 2 files changed, 32 insertions(+), 73 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 05252ac936a3..3168224bc104 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -225,15 +225,9 @@ struct wm_coeff_ctl_ops { struct snd_ctl_elem_info *uinfo); }; -struct wm_coeff { - struct device *dev; - struct list_head ctl_list; - struct regmap *regmap; -}; - struct wm_coeff_ctl { const char *name; - struct snd_card *card; + struct snd_soc_card *card; struct wm_adsp_alg_region region; struct wm_coeff_ctl_ops ops; struct wm_adsp *adsp; @@ -378,7 +372,6 @@ static int wm_coeff_info(struct snd_kcontrol *kcontrol, static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, const void *buf, size_t len) { - struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; struct wm_adsp_alg_region *region = &ctl->region; const struct wm_adsp_region *mem; @@ -401,7 +394,7 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, if (!scratch) return -ENOMEM; - ret = regmap_raw_write(wm_coeff->regmap, reg, scratch, + ret = regmap_raw_write(adsp->regmap, reg, scratch, ctl->len); if (ret) { adsp_err(adsp, "Failed to write %zu bytes to %x\n", @@ -434,7 +427,6 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, void *buf, size_t len) { - struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; struct wm_adsp_alg_region *region = &ctl->region; const struct wm_adsp_region *mem; @@ -457,7 +449,7 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, if (!scratch) return -ENOMEM; - ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len); + ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len); if (ret) { adsp_err(adsp, "Failed to read %zu bytes from %x\n", ctl->len, reg); @@ -481,37 +473,18 @@ static int wm_coeff_get(struct snd_kcontrol *kcontrol, return 0; } -static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff, - struct wm_coeff_ctl *ctl, - const struct snd_kcontrol_new *kctl) -{ - int ret; - struct snd_kcontrol *kcontrol; - - kcontrol = snd_ctl_new1(kctl, wm_coeff); - ret = snd_ctl_add(ctl->card, kcontrol); - if (ret < 0) { - dev_err(wm_coeff->dev, "Failed to add %s: %d\n", - kctl->name, ret); - return ret; - } - ctl->kcontrol = kcontrol; - return 0; -} - struct wmfw_ctl_work { - struct wm_coeff *wm_coeff; + struct wm_adsp *adsp; struct wm_coeff_ctl *ctl; struct work_struct work; }; -static int wmfw_add_ctl(struct wm_coeff *wm_coeff, - struct wm_coeff_ctl *ctl) +static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) { struct snd_kcontrol_new *kcontrol; int ret; - if (!wm_coeff || !ctl || !ctl->name || !ctl->card) + if (!ctl || !ctl->name || !ctl->card) return -EINVAL; kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); @@ -525,14 +498,17 @@ static int wmfw_add_ctl(struct wm_coeff *wm_coeff, kcontrol->put = wm_coeff_put; kcontrol->private_value = (unsigned long)ctl; - ret = wm_coeff_add_kcontrol(wm_coeff, - ctl, kcontrol); + ret = snd_soc_add_card_controls(ctl->card, + kcontrol, 1); if (ret < 0) goto err_kcontrol; kfree(kcontrol); - list_add(&ctl->list, &wm_coeff->ctl_list); + ctl->kcontrol = snd_soc_card_get_kcontrol(ctl->card, + ctl->name); + + list_add(&ctl->list, &adsp->ctl_list); return 0; err_kcontrol: @@ -753,13 +729,12 @@ out: return ret; } -static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) +static int wm_coeff_init_control_caches(struct wm_adsp *adsp) { struct wm_coeff_ctl *ctl; int ret; - list_for_each_entry(ctl, &wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &adsp->ctl_list, list) { if (!ctl->enabled || ctl->set) continue; ret = wm_coeff_read_control(ctl->kcontrol, @@ -772,13 +747,12 @@ static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) return 0; } -static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) +static int wm_coeff_sync_controls(struct wm_adsp *adsp) { struct wm_coeff_ctl *ctl; int ret; - list_for_each_entry(ctl, &wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &adsp->ctl_list, list) { if (!ctl->enabled) continue; if (ctl->set) { @@ -799,7 +773,7 @@ static void wm_adsp_ctl_work(struct work_struct *work) struct wmfw_ctl_work, work); - wmfw_add_ctl(ctl_work->wm_coeff, ctl_work->ctl); + wmfw_add_ctl(ctl_work->adsp, ctl_work->ctl); kfree(ctl_work); } @@ -842,7 +816,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, snprintf(name, PAGE_SIZE, "DSP%d %s %x", dsp->num, region_name, region->alg); - list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list_for_each_entry(ctl, &dsp->ctl_list, list) { if (!strcmp(ctl->name, name)) { if (!ctl->enabled) @@ -866,7 +840,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, ctl->set = 0; ctl->ops.xget = wm_coeff_get; ctl->ops.xput = wm_coeff_put; - ctl->card = codec->card->snd_card; + ctl->card = codec->card; ctl->adsp = dsp; ctl->len = region->len; @@ -882,7 +856,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, goto err_ctl_cache; } - ctl_work->wm_coeff = dsp->wm_coeff; + ctl_work->adsp = dsp; ctl_work->ctl = ctl; INIT_WORK(&ctl_work->work, wm_adsp_ctl_work); schedule_work(&ctl_work->work); @@ -1434,12 +1408,12 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, goto err; /* Initialize caches for enabled and unset controls */ - ret = wm_coeff_init_control_caches(dsp->wm_coeff); + ret = wm_coeff_init_control_caches(dsp); if (ret != 0) goto err; /* Sync set controls */ - ret = wm_coeff_sync_controls(dsp->wm_coeff); + ret = wm_coeff_sync_controls(dsp); if (ret != 0) goto err; @@ -1460,10 +1434,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); - list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - } break; default: @@ -1591,12 +1563,12 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, goto err; /* Initialize caches for enabled and unset controls */ - ret = wm_coeff_init_control_caches(dsp->wm_coeff); + ret = wm_coeff_init_control_caches(dsp); if (ret != 0) goto err; /* Sync set controls */ - ret = wm_coeff_sync_controls(dsp->wm_coeff); + ret = wm_coeff_sync_controls(dsp); if (ret != 0) goto err; @@ -1637,10 +1609,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } - list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - } while (!list_empty(&dsp->alg_regions)) { alg_region = list_first_entry(&dsp->alg_regions, @@ -1679,49 +1649,38 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) } INIT_LIST_HEAD(&adsp->alg_regions); - - adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff), - GFP_KERNEL); - if (!adsp->wm_coeff) - return -ENOMEM; - adsp->wm_coeff->regmap = adsp->regmap; - adsp->wm_coeff->dev = adsp->dev; - INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list); + INIT_LIST_HEAD(&adsp->ctl_list); if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); - goto out_coeff; + return ret; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", ret); - goto out_coeff; + return ret; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", ret); - goto out_coeff; + return ret; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", ret); - goto out_coeff; + return ret; } } return 0; - -out_coeff: - kfree(adsp->wm_coeff); - return ret; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 9f922c82536c..64087fb1cdec 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -57,7 +57,7 @@ struct wm_adsp { struct regulator *dvfs; - struct wm_coeff *wm_coeff; + struct list_head ctl_list; }; #define WM_ADSP1(wname, num) \ -- cgit v1.2.3-70-g09d2 From 1deb57042fe2bd14cd7d4687f3c9418d26862053 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Jul 2013 08:50:28 +0100 Subject: ASoC: bfin-ac97: Fix prototype error following AC'97 refactoring As part of the multiplatform refactoring for AC'97 the AC'97 bus ops were staticised meaning that the prototype (which was never needed) conflicts with the declaration causing build failures. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/blackfin/bf5xx-ac97.h | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 15c635e33f4d..0c3e22d90a8d 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -9,7 +9,6 @@ #ifndef _BF5XX_AC97_H #define _BF5XX_AC97_H -extern struct snd_ac97_bus_ops bf5xx_ac97_ops; extern struct snd_ac97 *ac97; /* Frame format in memory, only support stereo currently */ struct ac97_frame { -- cgit v1.2.3-70-g09d2 From 0d47acc4ffaa9b63d96183d69d38bdb388314d7d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 17 Jul 2013 17:20:19 +0100 Subject: ASoC: smdk_wm8994: Make driver name more unique Avoid collisions with other SMDK audio by using the CODEC name. Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 581ea4a06fc6..05c479cf5b40 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -200,7 +200,7 @@ MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); static struct platform_driver smdk_audio_driver = { .driver = { - .name = "smdk-audio", + .name = "smdk-audio-wm8894", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), }, @@ -212,4 +212,4 @@ module_platform_driver(smdk_audio_driver); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:smdk-audio"); +MODULE_ALIAS("platform:smdk-audio-wm8994"); -- cgit v1.2.3-70-g09d2 From f6ecf50b5e33119620b446dd0bce8b0a01a39669 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 26 Jul 2013 12:01:53 +0100 Subject: ASoC: smdk_wm8994: Configure DAI format at init time Initialise the DAI format from the data link, saving code and repeated work. Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994.c | 17 ++++------------- 1 file changed, 4 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 05c479cf5b40..a56117536c94 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -41,7 +41,6 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; unsigned int pll_out; int ret; @@ -54,18 +53,6 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, else pll_out = params_rate(params) * 256; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1, SMDK_WM8994_FREQ, pll_out); if (ret < 0) @@ -131,6 +118,8 @@ static struct snd_soc_dai_link smdk_dai[] = { .platform_name = "samsung-i2s.0", .codec_name = "wm8994-codec", .init = smdk_wm8994_init_paiftx, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &smdk_ops, }, { /* Sec_Fifo Playback i/f */ .name = "Sec_FIFO TX", @@ -139,6 +128,8 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8994-aif1", .platform_name = "samsung-i2s-sec", .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &smdk_ops, }, }; -- cgit v1.2.3-70-g09d2 From d30c148bb1cab23d3c330e6352b8d882575a0c3a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 26 Jul 2013 12:42:19 +0100 Subject: ASoC: smdk_wm8994: Configure the MCLK1 rate based on the board Make the code more generally applicable by refactoring so that the MCLK1 rate can be selected based on the compatible string provided by the device tree, allowing use on other boards which have different rates or use other information sources. Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994.c | 37 +++++++++++++++++++++++++++++-------- 1 file changed, 29 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index a56117536c94..5fd7a05a9b9e 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -11,6 +11,7 @@ #include #include #include +#include /* * Default CFG switch settings to use this driver: @@ -37,6 +38,15 @@ /* SMDK has a 16.934MHZ crystal attached to WM8994 */ #define SMDK_WM8994_FREQ 16934000 +struct smdk_wm8994_data { + int mclk1_rate; +}; + +/* Default SMDKs */ +static struct smdk_wm8994_data smdk_board_data = { + .mclk1_rate = SMDK_WM8994_FREQ, +}; + static int smdk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -141,15 +151,28 @@ static struct snd_soc_card smdk = { .num_links = ARRAY_SIZE(smdk_dai), }; +#ifdef CONFIG_OF +static const struct of_device_id samsung_wm8994_of_match[] = { + { .compatible = "samsung,smdk-wm8994", .data = &smdk_board_data }, + {}, +}; +MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); +#endif /* CONFIG_OF */ static int smdk_audio_probe(struct platform_device *pdev) { int ret; struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &smdk; + struct smdk_wm8994_data *board; + const struct of_device_id *id; card->dev = &pdev->dev; + board = devm_kzalloc(&pdev->dev, sizeof(*board), GFP_KERNEL); + if (!board) + return -ENOMEM; + if (np) { smdk_dai[0].cpu_dai_name = NULL; smdk_dai[0].cpu_of_node = of_parse_phandle(np, @@ -164,6 +187,12 @@ static int smdk_audio_probe(struct platform_device *pdev) smdk_dai[0].platform_of_node = smdk_dai[0].cpu_of_node; } + id = of_match_device(samsung_wm8994_of_match, &pdev->dev); + if (id) + *board = *((struct smdk_wm8994_data *)id->data); + + platform_set_drvdata(pdev, board); + ret = snd_soc_register_card(card); if (ret) @@ -181,14 +210,6 @@ static int smdk_audio_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_OF -static const struct of_device_id samsung_wm8994_of_match[] = { - { .compatible = "samsung,smdk-wm8994", }, - {}, -}; -MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); -#endif /* CONFIG_OF */ - static struct platform_driver smdk_audio_driver = { .driver = { .name = "smdk-audio-wm8894", -- cgit v1.2.3-70-g09d2 From 564c65049eddb1a95b48958080db97eda88c98dd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Jul 2013 17:13:55 +0200 Subject: ASoC: dapm: Move snd_soc_dapm_update from dapm context to card The update field of a DAPM context is only assigned while the card's dapm_mutex is locked, the field is also cleared again while the mutex is stil locked. So there will only ever be one DAPM context at a time with a non-NULL update field. So it is safe to move the update field from the DAPM context struct to the card struct. Doing so will allow further cleanups in this area. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 -- include/sound/soc.h | 1 + sound/soc/soc-dapm.c | 22 +++++++++++----------- 3 files changed, 12 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e77c6f5a8390..3397292d94c8 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -575,8 +575,6 @@ struct snd_soc_dapm_context { struct delayed_work delayed_work; unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ - struct snd_soc_dapm_update *update; - void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); diff --git a/include/sound/soc.h b/include/sound/soc.h index 6eabee7ec15a..b1e1f967ae1e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1042,6 +1042,7 @@ struct snd_soc_card { /* Generic DAPM context for the card */ struct snd_soc_dapm_context dapm; struct snd_soc_dapm_stats dapm_stats; + struct snd_soc_dapm_update *update; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_card_root; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 366daef006ed..7449e27bf133 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1425,7 +1425,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, static void dapm_widget_update(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_update *update = dapm->update; + struct snd_soc_dapm_update *update = dapm->card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; unsigned int wi; @@ -1959,9 +1959,9 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm->update = update; + card->update = update; ret = soc_dapm_mux_update_power(dapm, kcontrol, mux, e); - dapm->update = NULL; + card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -2002,9 +2002,9 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm->update = update; + card->update = update; ret = soc_dapm_mixer_update_power(dapm, kcontrol, connect); - dapm->update = NULL; + card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -2693,11 +2693,11 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2775,11 +2775,11 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.reg = e->reg; update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2928,11 +2928,11 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.reg = e->reg; update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); -- cgit v1.2.3-70-g09d2 From 95dd5cd6e16d86786f7dc9da404ae477403d8f83 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Jul 2013 17:13:56 +0200 Subject: ASoC: dapm: Pass card instead of dapm context to dapm_power_widgets() DAPM operations are always performed on the card as a whole. Yet (primarily for historic reasons) dapm_power_widgets() takes a DAPM context as its parameter. The DAPM context is mainly used to look up a pointer to the card. The same is true for a couple of functions that are being called from dapm_power_widgets(). This patch changes the signature of dapm_power_widgets() and a couple of related functions to take a snd_soc_card instead of a snd_soc_dapm_context. Some of the functions also use the DAPM's device to print error and debug messages. This can be a bit confusing though since this means the messages for all widgets, also those from other contexts, will be printed with that device. The patch updates those cases to use the device of the widget's DAPM context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 100 +++++++++++++++++++++++++-------------------------- 1 file changed, 49 insertions(+), 51 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7449e27bf133..5db8df2f8866 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1213,10 +1213,9 @@ static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget, list_add_tail(&new_widget->power_list, list); } -static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm, +static void dapm_seq_check_event(struct snd_soc_card *card, struct snd_soc_dapm_widget *w, int event) { - struct snd_soc_card *card = dapm->card; const char *ev_name; int power, ret; @@ -1254,22 +1253,21 @@ static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm, return; if (w->event && (w->event_flags & event)) { - pop_dbg(dapm->dev, card->pop_time, "pop test : %s %s\n", + pop_dbg(w->dapm->dev, card->pop_time, "pop test : %s %s\n", w->name, ev_name); trace_snd_soc_dapm_widget_event_start(w, event); ret = w->event(w, NULL, event); trace_snd_soc_dapm_widget_event_done(w, event); if (ret < 0) - dev_err(dapm->dev, "ASoC: %s: %s event failed: %d\n", + dev_err(w->dapm->dev, "ASoC: %s: %s event failed: %d\n", ev_name, w->name, ret); } } /* Apply the coalesced changes from a DAPM sequence */ -static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, +static void dapm_seq_run_coalesced(struct snd_soc_card *card, struct list_head *pending) { - struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; int reg, power; unsigned int value = 0; @@ -1292,13 +1290,13 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, if (power) value |= cur_mask; - pop_dbg(dapm->dev, card->pop_time, + pop_dbg(w->dapm->dev, card->pop_time, "pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n", w->name, reg, value, mask); /* Check for events */ - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_PRE_PMU); - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_PRE_PMD); + dapm_seq_check_event(card, w, SND_SOC_DAPM_PRE_PMU); + dapm_seq_check_event(card, w, SND_SOC_DAPM_PRE_PMD); } if (reg >= 0) { @@ -1308,7 +1306,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, w = list_first_entry(pending, struct snd_soc_dapm_widget, power_list); - pop_dbg(dapm->dev, card->pop_time, + pop_dbg(w->dapm->dev, card->pop_time, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); @@ -1316,8 +1314,8 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, } list_for_each_entry(w, pending, power_list) { - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_POST_PMU); - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_POST_PMD); + dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMU); + dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMD); } } @@ -1329,8 +1327,8 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, * Currently anything that requires more than a single write is not * handled. */ -static void dapm_seq_run(struct snd_soc_dapm_context *dapm, - struct list_head *list, int event, bool power_up) +static void dapm_seq_run(struct snd_soc_card *card, + struct list_head *list, int event, bool power_up) { struct snd_soc_dapm_widget *w, *n; LIST_HEAD(pending); @@ -1353,7 +1351,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, if (sort[w->id] != cur_sort || w->reg != cur_reg || w->dapm != cur_dapm || w->subseq != cur_subseq) { if (!list_empty(&pending)) - dapm_seq_run_coalesced(cur_dapm, &pending); + dapm_seq_run_coalesced(card, &pending); if (cur_dapm && cur_dapm->seq_notifier) { for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) @@ -1413,7 +1411,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, } if (!list_empty(&pending)) - dapm_seq_run_coalesced(cur_dapm, &pending); + dapm_seq_run_coalesced(card, &pending); if (cur_dapm && cur_dapm->seq_notifier) { for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) @@ -1423,9 +1421,9 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, } } -static void dapm_widget_update(struct snd_soc_dapm_context *dapm) +static void dapm_widget_update(struct snd_soc_card *card) { - struct snd_soc_dapm_update *update = dapm->card->update; + struct snd_soc_dapm_update *update = card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; unsigned int wi; @@ -1442,7 +1440,7 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm) if (w->event && (w->event_flags & SND_SOC_DAPM_PRE_REG)) { ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG); if (ret != 0) - dev_err(dapm->dev, "ASoC: %s DAPM pre-event failed: %d\n", + dev_err(w->dapm->dev, "ASoC: %s DAPM pre-event failed: %d\n", w->name, ret); } } @@ -1453,7 +1451,7 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm) ret = soc_widget_update_bits_locked(w, update->reg, update->mask, update->val); if (ret < 0) - dev_err(dapm->dev, "ASoC: %s DAPM update failed: %d\n", + dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", w->name, ret); for (wi = 0; wi < wlist->num_widgets; wi++) { @@ -1462,7 +1460,7 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm) if (w->event && (w->event_flags & SND_SOC_DAPM_POST_REG)) { ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG); if (ret != 0) - dev_err(dapm->dev, "ASoC: %s DAPM post-event failed: %d\n", + dev_err(w->dapm->dev, "ASoC: %s DAPM post-event failed: %d\n", w->name, ret); } } @@ -1627,9 +1625,8 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, * o Input pin to Output pin (bypass, sidetone) * o DAC to ADC (loopback). */ -static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) +static int dapm_power_widgets(struct snd_soc_card *card, int event) { - struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; struct snd_soc_dapm_context *d; LIST_HEAD(up_list); @@ -1711,29 +1708,29 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) trace_snd_soc_dapm_walk_done(card); /* Run all the bias changes in parallel */ - list_for_each_entry(d, &dapm->card->dapm_list, list) + list_for_each_entry(d, &card->dapm_list, list) async_schedule_domain(dapm_pre_sequence_async, d, &async_domain); async_synchronize_full_domain(&async_domain); list_for_each_entry(w, &down_list, power_list) { - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMD); + dapm_seq_check_event(card, w, SND_SOC_DAPM_WILL_PMD); } list_for_each_entry(w, &up_list, power_list) { - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU); + dapm_seq_check_event(card, w, SND_SOC_DAPM_WILL_PMU); } /* Power down widgets first; try to avoid amplifying pops. */ - dapm_seq_run(dapm, &down_list, event, false); + dapm_seq_run(card, &down_list, event, false); - dapm_widget_update(dapm); + dapm_widget_update(card); /* Now power up. */ - dapm_seq_run(dapm, &up_list, event, true); + dapm_seq_run(card, &up_list, event, true); /* Run all the bias changes in parallel */ - list_for_each_entry(d, &dapm->card->dapm_list, list) + list_for_each_entry(d, &card->dapm_list, list) async_schedule_domain(dapm_post_sequence_async, d, &async_domain); async_synchronize_full_domain(&async_domain); @@ -1744,7 +1741,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) d->stream_event(d, event); } - pop_dbg(dapm->dev, card->pop_time, + pop_dbg(card->dev, card->pop_time, "DAPM sequencing finished, waiting %dms\n", card->pop_time); pop_wait(card->pop_time); @@ -1917,14 +1914,14 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) #endif /* test and update the power status of a mux widget */ -static int soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, +static int soc_dapm_mux_update_power(struct snd_soc_card *card, struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &dapm->card->paths, list) { + list_for_each_entry(path, &card->paths, list) { if (path->kcontrol != kcontrol) continue; @@ -1946,7 +1943,7 @@ static int soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, } if (found) - dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP); return found; } @@ -1960,7 +1957,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); card->update = update; - ret = soc_dapm_mux_update_power(dapm, kcontrol, mux, e); + ret = soc_dapm_mux_update_power(card, kcontrol, mux, e); card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) @@ -1970,14 +1967,14 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); /* test and update the power status of a mixer or switch widget */ -static int soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, +static int soc_dapm_mixer_update_power(struct snd_soc_card *card, struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; int found = 0; /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &dapm->card->paths, list) { + list_for_each_entry(path, &card->paths, list) { if (path->kcontrol != kcontrol) continue; @@ -1989,7 +1986,7 @@ static int soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, } if (found) - dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP); return found; } @@ -2003,7 +2000,7 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); card->update = update; - ret = soc_dapm_mixer_update_power(dapm, kcontrol, connect); + ret = soc_dapm_mixer_update_power(card, kcontrol, connect); card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) @@ -2180,7 +2177,7 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) return 0; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + ret = dapm_power_widgets(dapm->card, SND_SOC_DAPM_STREAM_NOP); mutex_unlock(&dapm->card->dapm_mutex); return ret; } @@ -2545,12 +2542,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); */ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) { + struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; unsigned int val; - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); - list_for_each_entry(w, &dapm->card->widgets, list) + list_for_each_entry(w, &card->widgets, list) { if (w->new) continue; @@ -2560,7 +2558,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) sizeof(struct snd_kcontrol *), GFP_KERNEL); if (!w->kcontrols) { - mutex_unlock(&dapm->card->dapm_mutex); + mutex_unlock(&card->dapm_mutex); return -ENOMEM; } } @@ -2601,8 +2599,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) dapm_debugfs_add_widget(w); } - dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); - mutex_unlock(&dapm->card->dapm_mutex); + dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP); + mutex_unlock(&card->dapm_mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); @@ -2695,7 +2693,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, card->update = &update; - soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect); + soc_dapm_mixer_update_power(card, kcontrol, connect); card->update = NULL; } @@ -2777,7 +2775,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; - soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); + soc_dapm_mux_update_power(card, kcontrol, mux, e); card->update = NULL; } @@ -2832,7 +2830,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, change = widget->value != ucontrol->value.enumerated.item[0]; if (change) { widget->value = ucontrol->value.enumerated.item[0]; - soc_dapm_mux_update_power(widget->dapm, kcontrol, widget->value, e); + soc_dapm_mux_update_power(card, kcontrol, widget->value, e); } mutex_unlock(&card->dapm_mutex); @@ -2930,7 +2928,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; - soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); + soc_dapm_mux_update_power(card, kcontrol, mux, e); card->update = NULL; } @@ -3478,7 +3476,7 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, } } - dapm_power_widgets(&rtd->card->dapm, event); + dapm_power_widgets(rtd->card, event); } /** @@ -3747,7 +3745,7 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) if (dapm->bias_level == SND_SOC_BIAS_ON) snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE); - dapm_seq_run(dapm, &down_list, 0, false); + dapm_seq_run(card, &down_list, 0, false); if (dapm->bias_level == SND_SOC_BIAS_PREPARE) snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); -- cgit v1.2.3-70-g09d2 From eee5d7f99ae95059e1a3d1cfa2dea3ed8dbd94ac Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Jul 2013 17:13:57 +0200 Subject: ASoC: dapm: Add a helper to get the CODEC for DAPM kcontrol We use the same 3 lines to get the CODEC for a kcontrol in a quite a few places. This patch puts them into a common helper function. Having this encapsulated in a helper function will also make it more easier to eventually change the data layout of the kcontrol's private data. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 ++ sound/soc/codecs/tlv320aic3x.c | 7 +++---- sound/soc/codecs/twl6040.c | 4 +--- sound/soc/codecs/wm8903.c | 4 +--- sound/soc/codecs/wm8994.c | 4 +--- sound/soc/codecs/wm8995.c | 5 +---- sound/soc/codecs/wm_hubs.c | 8 ++------ sound/soc/soc-dapm.c | 40 +++++++++++++++++++++++----------------- 8 files changed, 34 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3397292d94c8..ebfae8f3fda7 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -427,6 +427,8 @@ void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list); +struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 1325c0c0df50..fec0db04262d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -138,8 +138,7 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -165,14 +164,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, mask <<= shift; val <<= shift; - change = snd_soc_test_bits(widget->codec, val, mask, reg); + change = snd_soc_test_bits(codec, val, mask, reg); if (change) { update.kcontrol = kcontrol; update.reg = reg; update.mask = mask; update.val = val; - snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect, + snd_soc_dapm_mixer_update_power(&codec->dapm, kcontrol, connect, &update); } diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 44621ddc332d..d6c5bf14179a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -437,9 +437,7 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index fa24cedee687..eebcb1da3b7b 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -364,9 +364,7 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); u16 reg; int ret; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ba832b77c543..eee2a01f2691 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1437,9 +1437,7 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *w = wlist->widgets[0]; - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 90a65c427541..da2899e6c401 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -549,12 +549,9 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source, static int wm8995_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *w = wlist->widgets[0]; - struct snd_soc_codec *codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; - codec = w->codec; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); wm8995_update_class_w(codec); return ret; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2d9e099415a5..8b50e5958de5 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -699,9 +699,7 @@ EXPORT_SYMBOL_GPL(wm_hubs_update_class_w); static int class_w_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); @@ -721,9 +719,7 @@ static int class_w_put_volsw(struct snd_kcontrol *kcontrol, static int class_w_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5db8df2f8866..b18ac5b1cc2e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -174,6 +174,17 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +/** + * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol + * @kcontrol: The kcontrol + */ +struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol) +{ + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + return wlist->widgets[0]->codec; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec); + static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; @@ -2617,8 +2628,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -2628,12 +2638,12 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, unsigned int invert = mc->invert; if (snd_soc_volsw_is_stereo(mc)) - dev_warn(widget->dapm->dev, + dev_warn(codec->dapm.dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); ucontrol->value.integer.value[0] = - (snd_soc_read(widget->codec, reg) >> shift) & mask; + (snd_soc_read(codec, reg) >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; @@ -2654,9 +2664,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw); int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; @@ -2670,7 +2678,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_update update; if (snd_soc_volsw_is_stereo(mc)) - dev_warn(widget->dapm->dev, + dev_warn(codec->dapm.dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); @@ -2684,7 +2692,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - change = snd_soc_test_bits(widget->codec, reg, mask, val); + change = snd_soc_test_bits(codec, reg, mask, val); if (change) { update.kcontrol = kcontrol; update.reg = reg; @@ -2715,12 +2723,11 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val; - val = snd_soc_read(widget->codec, e->reg); + val = snd_soc_read(codec, e->reg); ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask; if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = @@ -2765,7 +2772,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + change = snd_soc_test_bits(codec, e->reg, mask, val); if (change) { widget->value = val; @@ -2854,12 +2861,11 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val, mux; - reg_val = snd_soc_read(widget->codec, e->reg); + reg_val = snd_soc_read(codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; for (mux = 0; mux < e->max; mux++) { if (val == e->values[mux]) @@ -2918,7 +2924,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + change = snd_soc_test_bits(codec, e->reg, mask, val); if (change) { widget->value = val; -- cgit v1.2.3-70-g09d2 From e84357f7608f230b905acb18fe668609c9b811f0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Jul 2013 17:13:58 +0200 Subject: ASoC: dapm: Wrap kcontrol widget list access In preparation for adding additional per control data wrap all access to the widget list in helper functions. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 119 +++++++++++++++++++++++++++++++++------------------ 1 file changed, 78 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b18ac5b1cc2e..da35b10ce6d1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -174,14 +174,72 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +struct dapm_kcontrol_data { + struct snd_soc_dapm_widget_list wlist; +}; + +static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data; + + data = kzalloc(sizeof(*data) + sizeof(widget), GFP_KERNEL); + if (!data) { + dev_err(widget->dapm->dev, + "ASoC: can't allocate kcontrol data for %s\n", + widget->name); + return -ENOMEM; + } + + data->wlist.widgets[0] = widget; + data->wlist.num_widgets = 1; + + kcontrol->private_data = data; + + return 0; +} + +static void dapm_kcontrol_free(struct snd_kcontrol *kctl) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); + kfree(data); +} + +static struct snd_soc_dapm_widget_list *dapm_kcontrol_get_wlist( + const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + return &data->wlist; +} + +static int dapm_kcontrol_add_widget(struct snd_kcontrol *kcontrol, + struct snd_soc_dapm_widget *widget) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + struct dapm_kcontrol_data *new_data; + unsigned int n = data->wlist.num_widgets + 1; + + new_data = krealloc(data, sizeof(*data) + sizeof(widget) * n, + GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->wlist.widgets[n - 1] = widget; + data->wlist.num_widgets = n; + + kcontrol->private_data = data; + + return 0; +} + /** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol * @kcontrol: The kcontrol */ struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - return wlist->widgets[0]->codec; + return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->codec; } EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec); @@ -488,11 +546,6 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, return 0; } -static void dapm_kcontrol_free(struct snd_kcontrol *kctl) -{ - kfree(kctl->private_data); -} - /* * Determine if a kcontrol is shared. If it is, look it up. If it isn't, * create it. Either way, add the widget into the control's widget list @@ -506,9 +559,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, size_t prefix_len; int shared; struct snd_kcontrol *kcontrol; - struct snd_soc_dapm_widget_list *wlist; - int wlistentries; - size_t wlistsize; bool wname_in_long_name, kcname_in_long_name; char *long_name; const char *name; @@ -527,25 +577,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[kci], &kcontrol); - if (kcontrol) { - wlist = kcontrol->private_data; - wlistentries = wlist->num_widgets + 1; - } else { - wlist = NULL; - wlistentries = 1; - } - - wlistsize = sizeof(struct snd_soc_dapm_widget_list) + - wlistentries * sizeof(struct snd_soc_dapm_widget *); - wlist = krealloc(wlist, wlistsize, GFP_KERNEL); - if (wlist == NULL) { - dev_err(dapm->dev, "ASoC: can't allocate widget list for %s\n", - w->name); - return -ENOMEM; - } - wlist->num_widgets = wlistentries; - wlist->widgets[wlistentries - 1] = w; - if (!kcontrol) { if (shared) { wname_in_long_name = false; @@ -568,7 +599,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, kcname_in_long_name = false; break; default: - kfree(wlist); return -EINVAL; } } @@ -583,10 +613,8 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, long_name = kasprintf(GFP_KERNEL, "%s %s", w->name + prefix_len, w->kcontrol_news[kci].name); - if (long_name == NULL) { - kfree(wlist); + if (long_name == NULL) return -ENOMEM; - } name = long_name; } else if (wname_in_long_name) { @@ -597,21 +625,30 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, name = w->kcontrol_news[kci].name; } - kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], wlist, name, + kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], NULL, name, prefix); kcontrol->private_free = dapm_kcontrol_free; kfree(long_name); + + ret = dapm_kcontrol_data_alloc(w, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + return ret; + } + ret = snd_ctl_add(card, kcontrol); if (ret < 0) { dev_err(dapm->dev, "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", w->name, name, ret); - kfree(wlist); return ret; } + } else { + ret = dapm_kcontrol_add_widget(kcontrol, w); + if (ret) + return ret; } - kcontrol->private_data = wlist; w->kcontrols[kci] = kcontrol; path->kcontrol = kcontrol; @@ -1443,7 +1480,7 @@ static void dapm_widget_update(struct snd_soc_card *card) if (!update) return; - wlist = snd_kcontrol_chip(update->kcontrol); + wlist = dapm_kcontrol_get_wlist(update->kcontrol); for (wi = 0; wi < wlist->num_widgets; wi++) { w = wlist->widgets[wi]; @@ -2749,7 +2786,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = dapm_kcontrol_get_wlist(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct snd_soc_card *card = codec->card; @@ -2802,7 +2839,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = dapm_kcontrol_get_wlist(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; ucontrol->value.enumerated.item[0] = widget->value; @@ -2821,7 +2858,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt); int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = dapm_kcontrol_get_wlist(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct snd_soc_card *card = codec->card; @@ -2901,7 +2938,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_double); int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = dapm_kcontrol_get_wlist(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct snd_soc_card *card = codec->card; -- cgit v1.2.3-70-g09d2 From cf7c1de20c576477d42deae255cbc6e439bb5dc0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Jul 2013 17:13:59 +0200 Subject: ASoC: dapm: Move 'value' field from widget to control The 'value' field is really per control and not per widget. Currently it is only used for virtual MUXes, which only have one control per widget. So in that case there is not so much of a difference between whether it is stored per widget or per control. Moving the 'value' field from the widget to the control will allow us to use it also for cases where we have more than one control per widget. E.g. for mixers with multiple input controls. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 - sound/soc/soc-dapm.c | 53 +++++++++++++++++++++++++++--------------------- 2 files changed, 30 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ebfae8f3fda7..d7d26cc8e3fc 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -523,7 +523,6 @@ struct snd_soc_dapm_widget { /* dapm control */ int reg; /* negative reg = no direct dapm */ unsigned char shift; /* bits to shift */ - unsigned int value; /* widget current value */ unsigned int mask; /* non-shifted mask */ unsigned int on_val; /* on state value */ unsigned int off_val; /* off state value */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index da35b10ce6d1..bad6f6db74c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -175,6 +175,7 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( } struct dapm_kcontrol_data { + unsigned int value; struct snd_soc_dapm_widget_list wlist; }; @@ -233,6 +234,26 @@ static int dapm_kcontrol_add_widget(struct snd_kcontrol *kcontrol, return 0; } +static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + return data->value; +} + +static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, + unsigned int value) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + if (data->value == value) + return false; + + data->value = value; + + return true; +} + /** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol * @kcontrol: The kcontrol @@ -2786,9 +2807,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = dapm_kcontrol_get_wlist(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; @@ -2811,8 +2830,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, change = snd_soc_test_bits(codec, e->reg, mask, val); if (change) { - widget->value = val; - update.kcontrol = kcontrol; update.reg = e->reg; update.mask = mask; @@ -2839,11 +2856,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = dapm_kcontrol_get_wlist(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - - ucontrol->value.enumerated.item[0] = widget->value; - + ucontrol->value.enumerated.item[0] = dapm_kcontrol_get_value(kcontrol); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt); @@ -2858,10 +2871,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt); int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = dapm_kcontrol_get_wlist(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; + unsigned int value; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; @@ -2871,11 +2883,10 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - change = widget->value != ucontrol->value.enumerated.item[0]; - if (change) { - widget->value = ucontrol->value.enumerated.item[0]; - soc_dapm_mux_update_power(card, kcontrol, widget->value, e); - } + value = ucontrol->value.enumerated.item[0]; + change = dapm_kcontrol_set_value(kcontrol, value); + if (change) + soc_dapm_mux_update_power(card, kcontrol, value, e); mutex_unlock(&card->dapm_mutex); return change; @@ -2938,9 +2949,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_double); int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = dapm_kcontrol_get_wlist(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; @@ -2963,8 +2972,6 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, change = snd_soc_test_bits(codec, e->reg, mask, val); if (change) { - widget->value = val; - update.kcontrol = kcontrol; update.reg = e->reg; update.mask = mask; -- cgit v1.2.3-70-g09d2 From 5106b92f80a2cd37c52cffed80b4f5acfb77ccfd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Jul 2013 17:14:00 +0200 Subject: ASoC: dapm: Keep a list of paths per kcontrol Currently we store for each path which control (if any at all) is associated with that control. But we are only ever interested in the reverse relationship, i.e. we want to know all the paths a certain control is associated with. This is currently implemented by always iterating over all paths. This patch updates the code to keep a list for each control which contains all the paths that are associated with that control. This improves the run time of e.g. soc_dapm_mixer_update_power() and soc_dapm_mux_update_power() from O(n) (with n being the number of paths for the card) to O(1). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 +- sound/soc/soc-dapm.c | 40 ++++++++++++++++++++++++++++------------ 2 files changed, 29 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index d7d26cc8e3fc..693c75bbd5d1 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -490,7 +490,6 @@ struct snd_soc_dapm_path { /* source (input) and sink (output) widgets */ struct snd_soc_dapm_widget *source; struct snd_soc_dapm_widget *sink; - struct snd_kcontrol *kcontrol; /* status */ u32 connect:1; /* source and sink widgets are connected */ @@ -503,6 +502,7 @@ struct snd_soc_dapm_path { struct list_head list_source; struct list_head list_sink; + struct list_head list_kcontrol; struct list_head list; }; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bad6f6db74c9..b779d36d5b3a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -176,6 +176,7 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( struct dapm_kcontrol_data { unsigned int value; + struct list_head paths; struct snd_soc_dapm_widget_list wlist; }; @@ -194,6 +195,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->wlist.widgets[0] = widget; data->wlist.num_widgets = 1; + INIT_LIST_HEAD(&data->paths); kcontrol->private_data = data; @@ -234,6 +236,26 @@ static int dapm_kcontrol_add_widget(struct snd_kcontrol *kcontrol, return 0; } +static void dapm_kcontrol_add_path(const struct snd_kcontrol *kcontrol, + struct snd_soc_dapm_path *path) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + list_add_tail(&path->list_kcontrol, &data->paths); +} + +static struct list_head *dapm_kcontrol_get_path_list( + const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + return &data->paths; +} + +#define dapm_kcontrol_for_each_path(path, kcontrol) \ + list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \ + list_kcontrol) + static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); @@ -671,7 +693,7 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, } w->kcontrols[kci] = kcontrol; - path->kcontrol = kcontrol; + dapm_kcontrol_add_path(kcontrol, path); return 0; } @@ -691,7 +713,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) continue; if (w->kcontrols[i]) { - path->kcontrol = w->kcontrols[i]; + dapm_kcontrol_add_path(w->kcontrols[i], path); continue; } @@ -730,7 +752,7 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return ret; list_for_each_entry(path, &w->sources, list_sink) - path->kcontrol = w->kcontrols[0]; + dapm_kcontrol_add_path(w->kcontrols[0], path); return 0; } @@ -1990,10 +2012,7 @@ static int soc_dapm_mux_update_power(struct snd_soc_card *card, int found = 0; /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &card->paths, list) { - if (path->kcontrol != kcontrol) - continue; - + dapm_kcontrol_for_each_path(path, kcontrol) { if (!path->name || !e->texts[mux]) continue; @@ -2043,11 +2062,7 @@ static int soc_dapm_mixer_update_power(struct snd_soc_card *card, int found = 0; /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &card->paths, list) { - if (path->kcontrol != kcontrol) - continue; - - /* found, now check type */ + dapm_kcontrol_for_each_path(path, kcontrol) { found = 1; path->connect = connect; dapm_mark_dirty(path->source, "mixer connection"); @@ -2152,6 +2167,7 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) { list_del(&path->list_sink); list_del(&path->list_source); + list_del(&path->list_kcontrol); list_del(&path->list); kfree(path); } -- cgit v1.2.3-70-g09d2 From de9ba98b6d2629f53fd271a973176c2fa9736d9c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Jul 2013 17:14:01 +0200 Subject: ASoC: dapm: Make widget power register settings more flexible Currently the DAPM code is limited to only setting or clearing a single bit in a register to power a widget up or down. This patch extends the DAPM code to be more flexible in that regard and allow widgets to use arbitrary values to be used to put a widget in either on or off state. Since the snd_soc_dapm_widget struct already contains a on_val and off_val field no additional fields need to be added and in fact the invert field can even be removed. Also the generated code is slightly smaller. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 163 +++++++++++++++++++++++++++-------------------- sound/soc/soc-dapm.c | 34 ++++------ 2 files changed, 106 insertions(+), 91 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 693c75bbd5d1..3575721a955d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -70,121 +70,144 @@ struct device; .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD} +#define SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert) \ + .reg = wreg, .mask = 1, .shift = wshift, \ + .on_val = winvert ? 0 : 1, .off_val = winvert ? 1 : 0 + /* path domain */ #define SND_SOC_DAPM_PGA(wname, wreg, wshift, winvert,\ wcontrols, wncontrols) \ -{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols} +{ .id = snd_soc_dapm_pga, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = wncontrols} #define SND_SOC_DAPM_OUT_DRV(wname, wreg, wshift, winvert,\ wcontrols, wncontrols) \ -{ .id = snd_soc_dapm_out_drv, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols} +{ .id = snd_soc_dapm_out_drv, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = wncontrols} #define SND_SOC_DAPM_MIXER(wname, wreg, wshift, winvert, \ wcontrols, wncontrols)\ -{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols} +{ .id = snd_soc_dapm_mixer, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = wncontrols} #define SND_SOC_DAPM_MIXER_NAMED_CTL(wname, wreg, wshift, winvert, \ wcontrols, wncontrols)\ -{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \ - .shift = wshift, .invert = winvert, .kcontrol_news = wcontrols, \ - .num_kcontrols = wncontrols} +{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = wncontrols} #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \ -{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = NULL, .num_kcontrols = 0} +{ .id = snd_soc_dapm_micbias, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = NULL, .num_kcontrols = 0} #define SND_SOC_DAPM_SWITCH(wname, wreg, wshift, winvert, wcontrols) \ -{ .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1} +{ .id = snd_soc_dapm_switch, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = 1} #define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \ -{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1} +{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, \ + .kcontrol_news = wcontrols, .num_kcontrols = 1} #define SND_SOC_DAPM_VIRT_MUX(wname, wreg, wshift, winvert, wcontrols) \ -{ .id = snd_soc_dapm_virt_mux, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1} +{ .id = snd_soc_dapm_virt_mux, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = 1} #define SND_SOC_DAPM_VALUE_MUX(wname, wreg, wshift, winvert, wcontrols) \ -{ .id = snd_soc_dapm_value_mux, .name = wname, .reg = wreg, \ - .shift = wshift, .invert = winvert, .kcontrol_news = wcontrols, \ - .num_kcontrols = 1} +{ .id = snd_soc_dapm_value_mux, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = 1} /* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ #define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\ wcontrols) \ -{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} +{ .id = snd_soc_dapm_pga, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} #define SOC_MIXER_ARRAY(wname, wreg, wshift, winvert, \ wcontrols)\ -{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} +{ .id = snd_soc_dapm_mixer, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} #define SOC_MIXER_NAMED_CTL_ARRAY(wname, wreg, wshift, winvert, \ wcontrols)\ -{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \ - .shift = wshift, .invert = winvert, .kcontrol_news = wcontrols, \ - .num_kcontrols = ARRAY_SIZE(wcontrols)} +{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} /* path domain with event - event handler must return 0 for success */ #define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \ wncontrols, wevent, wflags) \ -{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \ +{ .id = snd_soc_dapm_pga, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \ .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_OUT_DRV_E(wname, wreg, wshift, winvert, wcontrols, \ wncontrols, wevent, wflags) \ -{ .id = snd_soc_dapm_out_drv, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \ +{ .id = snd_soc_dapm_out_drv, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \ .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_MIXER_E(wname, wreg, wshift, winvert, wcontrols, \ wncontrols, wevent, wflags) \ -{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \ +{ .id = snd_soc_dapm_mixer, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \ .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_MIXER_NAMED_CTL_E(wname, wreg, wshift, winvert, \ wcontrols, wncontrols, wevent, wflags) \ -{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, \ +{ .id = snd_soc_dapm_mixer, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, \ .num_kcontrols = wncontrols, .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \ wevent, wflags) \ -{ .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1, \ +{ .id = snd_soc_dapm_switch, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = 1, \ .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, \ wevent, wflags) \ -{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1, \ +{ .id = snd_soc_dapm_mux, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = 1, \ .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_VIRT_MUX_E(wname, wreg, wshift, winvert, wcontrols, \ wevent, wflags) \ -{ .id = snd_soc_dapm_virt_mux, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1, \ +{ .id = snd_soc_dapm_virt_mux, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = 1, \ .event = wevent, .event_flags = wflags} /* additional sequencing control within an event type */ #define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, \ wevent, wflags) \ -{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .event = wevent, .event_flags = wflags, \ +{ .id = snd_soc_dapm_pga, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .event = wevent, .event_flags = wflags, \ .subseq = wsubseq} #define SND_SOC_DAPM_SUPPLY_S(wname, wsubseq, wreg, wshift, winvert, wevent, \ wflags) \ -{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ - .shift = wshift, .invert = winvert, .event = wevent, \ - .event_flags = wflags, .subseq = wsubseq} +{ .id = snd_soc_dapm_supply, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .event = wevent, .event_flags = wflags, .subseq = wsubseq} /* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ #define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \ wevent, wflags) \ -{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ +{ .id = snd_soc_dapm_pga, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ .event = wevent, .event_flags = wflags} #define SOC_MIXER_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \ wevent, wflags) \ -{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ +{ .id = snd_soc_dapm_mixer, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ .event = wevent, .event_flags = wflags} #define SOC_MIXER_NAMED_CTL_E_ARRAY(wname, wreg, wshift, winvert, \ wcontrols, wevent, wflags) \ -{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = wcontrols, \ - .num_kcontrols = ARRAY_SIZE(wcontrols), .event = wevent, .event_flags = wflags} +{ .id = snd_soc_dapm_mixer, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ + .event = wevent, .event_flags = wflags} /* events that are pre and post DAPM */ #define SND_SOC_DAPM_PRE(wname, wevent) \ @@ -199,35 +222,36 @@ struct device; /* stream domain */ #define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \ { .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ - .reg = wreg, .shift = wshift, .invert = winvert } + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } #define SND_SOC_DAPM_AIF_IN_E(wname, stname, wslot, wreg, wshift, winvert, \ wevent, wflags) \ { .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ - .reg = wreg, .shift = wshift, .invert = winvert, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .event = wevent, .event_flags = wflags } #define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \ { .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ - .reg = wreg, .shift = wshift, .invert = winvert } + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } #define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wslot, wreg, wshift, winvert, \ wevent, wflags) \ { .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ - .reg = wreg, .shift = wshift, .invert = winvert, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .event = wevent, .event_flags = wflags } #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ -{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ - .shift = wshift, .invert = winvert} +{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert) } #define SND_SOC_DAPM_DAC_E(wname, stname, wreg, wshift, winvert, \ wevent, wflags) \ -{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ - .shift = wshift, .invert = winvert, \ +{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .event = wevent, .event_flags = wflags} + #define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \ -{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ - .shift = wshift, .invert = winvert} +{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } #define SND_SOC_DAPM_ADC_E(wname, stname, wreg, wshift, winvert, \ wevent, wflags) \ -{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ - .shift = wshift, .invert = winvert, \ +{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \ { .id = snd_soc_dapm_clock_supply, .name = wname, \ @@ -241,14 +265,14 @@ struct device; .on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} #define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \ -{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ - .shift = wshift, .invert = winvert, .event = wevent, \ - .event_flags = wflags} +{ .id = snd_soc_dapm_supply, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay, wflags) \ { .id = snd_soc_dapm_regulator_supply, .name = wname, \ .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ - .invert = wflags} + .on_val = wflags} /* dapm kcontrol types */ @@ -527,7 +551,6 @@ struct snd_soc_dapm_widget { unsigned int on_val; /* on state value */ unsigned int off_val; /* off state value */ unsigned char power:1; /* block power status */ - unsigned char invert:1; /* invert the power bit */ unsigned char active:1; /* active stream on DAC, ADC's */ unsigned char connected:1; /* connected codec pin */ unsigned char new:1; /* cnew complete */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b779d36d5b3a..59bcc66358ca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1122,7 +1122,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, int ret; if (SND_SOC_DAPM_EVENT_ON(event)) { - if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, @@ -1132,7 +1132,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, return regulator_enable(w->regulator); } else { - if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, @@ -1360,26 +1360,21 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, struct list_head *pending) { struct snd_soc_dapm_widget *w; - int reg, power; + int reg; unsigned int value = 0; unsigned int mask = 0; - unsigned int cur_mask; reg = list_first_entry(pending, struct snd_soc_dapm_widget, power_list)->reg; list_for_each_entry(w, pending, power_list) { - cur_mask = 1 << w->shift; BUG_ON(reg != w->reg); - if (w->invert) - power = !w->power; + mask |= w->mask << w->shift; + if (w->power) + value |= w->on_val << w->shift; else - power = w->power; - - mask |= cur_mask; - if (power) - value |= cur_mask; + value |= w->off_val << w->shift; pop_dbg(w->dapm->dev, card->pop_time, "pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n", @@ -1867,8 +1862,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (w->reg >= 0) ret += snprintf(buf + ret, PAGE_SIZE - ret, - " - R%d(0x%x) bit %d", - w->reg, w->reg, w->shift); + " - R%d(0x%x) mask 0x%x", + w->reg, w->reg, w->mask << w->shift); ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); @@ -2669,12 +2664,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) /* Read the initial power state from the device */ if (w->reg >= 0) { - val = soc_widget_read(w, w->reg); - val &= 1 << w->shift; - if (w->invert) - val = !val; - - if (val) + val = soc_widget_read(w, w->reg) >> w->shift; + val &= w->mask; + if (val == w->on_val) w->power = 1; } @@ -3093,7 +3085,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, return NULL; } - if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, -- cgit v1.2.3-70-g09d2 From 2553628e1973709bf378320ecffd3e4fb34458db Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Jul 2013 17:14:02 +0200 Subject: ASoC: dapm: Add snd_soc_dapm_add_path() helper function snd_soc_dapm_add_path() is similar to snd_soc_dapm_add_route() except that it expects the pointer to the source and sink widgets instead of their names. This allows us to simplify the case where we already have a pointer to widgets. (E.g. as we have in snd_soc_dapm_link_dai_widgets()). snd_soc_dapm_add_route() will be updated to just look up the widget and then use snd_soc_dapm_add_path() to handle everything else. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 151 +++++++++++++++++++++++++++------------------------ 1 file changed, 81 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 59bcc66358ca..b811a27bf21a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2263,64 +2263,14 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); -static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_route *route) +static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink, + const char *control, + int (*connected)(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink)) { struct snd_soc_dapm_path *path; - struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; - struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL; - const char *sink; - const char *control = route->control; - const char *source; - char prefixed_sink[80]; - char prefixed_source[80]; - int ret = 0; - - if (dapm->codec && dapm->codec->name_prefix) { - snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", - dapm->codec->name_prefix, route->sink); - sink = prefixed_sink; - snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", - dapm->codec->name_prefix, route->source); - source = prefixed_source; - } else { - sink = route->sink; - source = route->source; - } - - /* - * find src and dest widgets over all widgets but favor a widget from - * current DAPM context - */ - list_for_each_entry(w, &dapm->card->widgets, list) { - if (!wsink && !(strcmp(w->name, sink))) { - wtsink = w; - if (w->dapm == dapm) - wsink = w; - continue; - } - if (!wsource && !(strcmp(w->name, source))) { - wtsource = w; - if (w->dapm == dapm) - wsource = w; - } - } - /* use widget from another DAPM context if not found from this */ - if (!wsink) - wsink = wtsink; - if (!wsource) - wsource = wtsource; - - if (wsource == NULL) { - dev_err(dapm->dev, "ASoC: no source widget found for %s\n", - route->source); - return -ENODEV; - } - if (wsink == NULL) { - dev_err(dapm->dev, "ASoC: no sink widget found for %s\n", - route->sink); - return -ENODEV; - } + int ret; path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL); if (!path) @@ -2328,7 +2278,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, path->source = wsource; path->sink = wsink; - path->connected = route->connected; + path->connected = connected; INIT_LIST_HEAD(&path->list); INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); @@ -2414,11 +2364,77 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(wsink, "Route added"); return 0; +err: + kfree(path); + return ret; +} +static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route) +{ + struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; + struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL; + const char *sink; + const char *source; + char prefixed_sink[80]; + char prefixed_source[80]; + int ret; + + if (dapm->codec && dapm->codec->name_prefix) { + snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", + dapm->codec->name_prefix, route->sink); + sink = prefixed_sink; + snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", + dapm->codec->name_prefix, route->source); + source = prefixed_source; + } else { + sink = route->sink; + source = route->source; + } + + /* + * find src and dest widgets over all widgets but favor a widget from + * current DAPM context + */ + list_for_each_entry(w, &dapm->card->widgets, list) { + if (!wsink && !(strcmp(w->name, sink))) { + wtsink = w; + if (w->dapm == dapm) + wsink = w; + continue; + } + if (!wsource && !(strcmp(w->name, source))) { + wtsource = w; + if (w->dapm == dapm) + wsource = w; + } + } + /* use widget from another DAPM context if not found from this */ + if (!wsink) + wsink = wtsink; + if (!wsource) + wsource = wtsource; + + if (wsource == NULL) { + dev_err(dapm->dev, "ASoC: no source widget found for %s\n", + route->source); + return -ENODEV; + } + if (wsink == NULL) { + dev_err(dapm->dev, "ASoC: no sink widget found for %s\n", + route->sink); + return -ENODEV; + } + + ret = snd_soc_dapm_add_path(dapm, wsource, wsink, route->control, + route->connected); + if (ret) + goto err; + + return 0; err: dev_warn(dapm->dev, "ASoC: no dapm match for %s --> %s --> %s\n", - source, control, sink); - kfree(path); + source, route->control, sink); return ret; } @@ -3421,9 +3437,6 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) { struct snd_soc_dapm_widget *dai_w, *w; struct snd_soc_dai *dai; - struct snd_soc_dapm_route r; - - memset(&r, 0, sizeof(r)); /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { @@ -3456,23 +3469,21 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) if (dai->driver->playback.stream_name && strstr(w->sname, dai->driver->playback.stream_name)) { - r.source = dai->playback_widget->name; - r.sink = w->name; dev_dbg(dai->dev, "%s -> %s\n", - r.source, r.sink); + dai->playback_widget->name, w->name); - snd_soc_dapm_add_route(w->dapm, &r); + snd_soc_dapm_add_path(w->dapm, + dai->playback_widget, w, NULL, NULL); } if (dai->driver->capture.stream_name && strstr(w->sname, dai->driver->capture.stream_name)) { - r.source = w->name; - r.sink = dai->capture_widget->name; dev_dbg(dai->dev, "%s -> %s\n", - r.source, r.sink); + w->name, dai->capture_widget->name); - snd_soc_dapm_add_route(w->dapm, &r); + snd_soc_dapm_add_path(w->dapm, w, + dai->capture_widget, NULL, NULL); } } } -- cgit v1.2.3-70-g09d2 From 39eb5fd13dff8d3d04489fe3f59e0d22bf89041e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Jul 2013 17:14:03 +0200 Subject: ASoC: dapm: Delay w->power update until the changes are written Wait with updating the widgets power field until the changes are actually written to the hardware in dapm_seq_run_coalesced(). This will allow us to query the current hardware state between calling dapm_power_one_widget() and actually writing the new power state to hardware. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b811a27bf21a..9abb3b21f1fd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -293,6 +293,7 @@ static void dapm_reset(struct snd_soc_card *card) memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); list_for_each_entry(w, &card->widgets, list) { + w->new_power = w->power; w->power_checked = false; w->inputs = -1; w->outputs = -1; @@ -1340,7 +1341,7 @@ static void dapm_seq_check_event(struct snd_soc_card *card, return; } - if (w->power != power) + if (w->new_power != power) return; if (w->event && (w->event_flags & event)) { @@ -1369,6 +1370,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, list_for_each_entry(w, pending, power_list) { BUG_ON(reg != w->reg); + w->power = w->new_power; mask |= w->mask << w->shift; if (w->power) @@ -1676,8 +1678,6 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, dapm_seq_insert(w, up_list, true); else dapm_seq_insert(w, down_list, false); - - w->power = power; } static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, @@ -1752,7 +1752,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) break; } - if (w->power) { + if (w->new_power) { d = w->dapm; /* Supplies and micbiases only bring the -- cgit v1.2.3-70-g09d2 From 113591e477acb6b6dbc186ad2ee29a2502e68c33 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 30 Jul 2013 11:18:52 +0100 Subject: ASoC: uda134x: fix codec driver by converting to DAPM For some reason, the DAC/ADCs are not being powered up when I try and use the UDA1341 driver; this used to work. Looking back in the git history, I don't see anything obvious which would cause this regression. However, from dumping the register writes, it seems that the codec is powered down, and nothing calls set_bias_level to wake the codec up. Moreover, this driver hasn't had DAPM support added to it, which prevents platform drivers from taking advantage of DAPMs facilities. So, let's add DAPM support to the driver. As we move the power control for the DAC/ADC into DAPM, we no longer need it in set_bias_level() - this function just becomes a way to manipulate the power control and sync the register cache with the hardware at the appropriate point. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/codecs/uda134x.c | 88 ++++++++++++++++++++++++++++------------------ 1 file changed, 53 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 6d0aa44c3757..c94d4c1e3dac 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -325,7 +325,6 @@ static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, static int uda134x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u8 reg; struct uda134x_platform_data *pd = codec->control_data; int i; u8 *cache = codec->reg_cache; @@ -334,23 +333,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - /* ADC, DAC on */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg | 0x03); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_PREPARE: /* power on */ @@ -362,23 +344,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - /* ADC, DAC power off */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03)); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_OFF: /* power off */ @@ -450,6 +415,37 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; +/* UDA1341 has the DAC/ADC power down in STATUS1 */ +static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0), +}; + +/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */ +static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0), +}; + +/* Common DAPM widgets */ +static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("VINL1"), + SND_SOC_DAPM_INPUT("VINR1"), + SND_SOC_DAPM_INPUT("VINL2"), + SND_SOC_DAPM_INPUT("VINR2"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route uda134x_dapm_routes[] = { + { "ADC", NULL, "VINL1" }, + { "ADC", NULL, "VINR1" }, + { "ADC", NULL, "VINL2" }, + { "ADC", NULL, "VINR2" }, + { "VOUTL", NULL, "DAC" }, + { "VOUTR", NULL, "DAC" }, +}; + static const struct snd_soc_dai_ops uda134x_dai_ops = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, @@ -485,6 +481,8 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; struct uda134x_platform_data *pd = codec->card->dev->platform_data; + const struct snd_soc_dapm_widget *widgets; + unsigned num_widgets; int ret; @@ -526,6 +524,22 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) else uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (pd->model == UDA134X_UDA1341) { + widgets = uda1341_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1341_dapm_widgets); + } else { + widgets = uda1340_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1340_dapm_widgets); + } + + ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets); + if (ret) { + printk(KERN_ERR "%s failed to register dapm controls: %d", + __func__, ret); + kfree(uda134x); + return ret; + } + switch (pd->model) { case UDA134X_UDA1340: case UDA134X_UDA1344: @@ -599,6 +613,10 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = { .read = uda134x_read_reg_cache, .write = uda134x_write, .set_bias_level = uda134x_set_bias_level, + .dapm_widgets = uda134x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets), + .dapm_routes = uda134x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes), }; static int uda134x_codec_probe(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From 0890c2b7be08b0928f7f507e371918205a0312f7 Mon Sep 17 00:00:00 2001 From: Richard Genoud Date: Tue, 30 Jul 2013 11:59:45 +0200 Subject: ASoC: wm8731: add rates constraints Depending on the mclk (or crystal) selected, the wm8731 codec have some constraints on its data sampling rates: e.g. with a 12.288MHz or 18.432MHz crystal, the authorized rates are 8KHz, 32KHz, 48KHz and 96KHz. Signed-off-by: Richard Genoud Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 60 ++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 57 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5276062d6c79..456bb8c6d759 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -45,6 +45,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { struct wm8731_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; + const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; int sysclk_type; int playback_fs; @@ -290,6 +291,36 @@ static const struct _coeff_div coeff_div[] = { {12000000, 88200, 136, 0xf, 0x1, 0x1}, }; +/* rates constraints */ +static const unsigned int wm8731_rates_12000000[] = { + 8000, 32000, 44100, 48000, 96000, 88200, +}; + +static const unsigned int wm8731_rates_12288000_18432000[] = { + 8000, 32000, 48000, 96000, +}; + +static const unsigned int wm8731_rates_11289600_16934400[] = { + 8000, 44100, 88200, +}; + +static const struct snd_pcm_hw_constraint_list wm8731_constraints_12000000 = { + .list = wm8731_rates_12000000, + .count = ARRAY_SIZE(wm8731_rates_12000000), +}; + +static const +struct snd_pcm_hw_constraint_list wm8731_constraints_12288000_18432000 = { + .list = wm8731_rates_12288000_18432000, + .count = ARRAY_SIZE(wm8731_rates_12288000_18432000), +}; + +static const +struct snd_pcm_hw_constraint_list wm8731_constraints_11289600_16934400 = { + .list = wm8731_rates_11289600_16934400, + .count = ARRAY_SIZE(wm8731_rates_11289600_16934400), +}; + static inline int get_coeff(int mclk, int rate) { int i; @@ -362,17 +393,26 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, } switch (freq) { - case 11289600: + case 0: + wm8731->constraints = NULL; + break; case 12000000: + wm8731->constraints = &wm8731_constraints_12000000; + break; case 12288000: - case 16934400: case 18432000: - wm8731->sysclk = freq; + wm8731->constraints = &wm8731_constraints_12288000_18432000; + break; + case 16934400: + case 11289600: + wm8731->constraints = &wm8731_constraints_11289600_16934400; break; default: return -EINVAL; } + wm8731->sysclk = freq; + snd_soc_dapm_sync(&codec->dapm); return 0; @@ -475,12 +515,26 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, return 0; } +static int wm8731_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(dai->codec); + + if (wm8731->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + wm8731->constraints); + + return 0; +} + #define WM8731_RATES SNDRV_PCM_RATE_8000_96000 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm8731_dai_ops = { + .startup = wm8731_startup, .hw_params = wm8731_hw_params, .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, -- cgit v1.2.3-70-g09d2 From 610d80eaa987e7b1a2d07ee800c9722e227a3b47 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 30 Jul 2013 13:34:09 +0200 Subject: ASoC: bf5xx-ac97: Fix compile error with SND_BF5XX_HAVE_COLD_RESET MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit If CONFIG_SND_BF5XX_HAVE_COLD_RESET is enabled building the blackfin ac97 driver fails with the following compile error: sound/soc/blackfin/bf5xx-ac97.c: In function ‘asoc_bfin_ac97_probe’: sound/soc/blackfin/bf5xx-ac97.c:297: error: expected ‘;’ before ‘{’ token sound/soc/blackfin/bf5xx-ac97.c:302: error: label ‘gpio_err’ used but not defined The issue was introduced in commit 6dab2fd7 ("ASoC: bf5xx-ac97: Convert to devm_gpio_request_one()"). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index efb1daecd0dd..e82eb373a731 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -294,11 +294,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) /* Request PB3 as reset pin */ ret = devm_gpio_request_one(&pdev->dev, CONFIG_SND_BF5XX_RESET_GPIO_NUM, - GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET") { + GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET"); + if (ret) { dev_err(&pdev->dev, "Failed to request GPIO_%d for reset: %d\n", CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); - goto gpio_err; + return ret; } #endif -- cgit v1.2.3-70-g09d2 From 50b4dc690a5c6ffed4c528829cf18f77e5af98bd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 30 Jul 2013 13:34:10 +0200 Subject: ASoC: bf5xx-ac97: Remove unused extern declaration The blackfin ac97 driver never defines nor uses a global ac97 struct. So remove the extern declaration for it. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.h | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 0c3e22d90a8d..a680fdc9bb42 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -9,7 +9,6 @@ #ifndef _BF5XX_AC97_H #define _BF5XX_AC97_H -extern struct snd_ac97 *ac97; /* Frame format in memory, only support stereo currently */ struct ac97_frame { u16 ac97_tag; /* slot 0 */ -- cgit v1.2.3-70-g09d2 From d2ee88d0aaacac664aff6ca5fc0bd7705d8f2414 Mon Sep 17 00:00:00 2001 From: Ralf Baechle Date: Wed, 31 Jul 2013 10:15:19 +0200 Subject: ASoC: au1x: Fix build MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit d8b51c11ff5a70244753ba60abfd47088cf4dcd4 [ASoC: ac97c: Use module_platform_driver()] broke the build: CC sound/soc/au1x/ac97c.o /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected identifier or ‘(’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: pasting "__initcall_" and "&" does not give a valid preprocessing token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected identifier or ‘(’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: pasting "__exitcall_" and "&" does not give a valid preprocessing token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:334:31: warning: ‘au1xac97c_driver’ defined but not used [-Wunused-variable] make[5]: *** [sound/soc/au1x/ac97c.o] Error 1 make[4]: *** [sound/soc/au1x] Error 2 make[3]: *** [sound/soc] Error 2 Signed-off-by: Ralf Baechle Signed-off-by: Mark Brown --- sound/soc/au1x/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d6f7694fcad4..c8a2de103c5f 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); -- cgit v1.2.3-70-g09d2 From 46a02c978fbc79de856d0fe7a8c1d4fc620796e0 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 31 Jul 2013 11:52:44 +0300 Subject: ASoC: dapm: using freed pointer in dapm_kcontrol_add_widget() There is a typo here so we end up using the old freed pointer instead of the newly allocated one. (If the "n" is zero then the code works, obviously). Signed-off-by: Dan Carpenter Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9abb3b21f1fd..d74c3560d556 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -225,13 +225,13 @@ static int dapm_kcontrol_add_widget(struct snd_kcontrol *kcontrol, new_data = krealloc(data, sizeof(*data) + sizeof(widget) * n, GFP_KERNEL); - if (!data) + if (!new_data) return -ENOMEM; - data->wlist.widgets[n - 1] = widget; - data->wlist.num_widgets = n; + new_data->wlist.widgets[n - 1] = widget; + new_data->wlist.num_widgets = n; - kcontrol->private_data = data; + kcontrol->private_data = new_data; return 0; } -- cgit v1.2.3-70-g09d2 From db5ff9541b61ef8394bad0fb05508921b8c5b17b Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 31 Jul 2013 20:07:05 +0800 Subject: ASoC: spdif: Add S20_3LE and S24_LE support for dummy codec drivers Generally, S/PDIF supports 20bit and optional 24bit samples. Thus add these two formats for the dummy codec drivers. If one S/PDIF controller has its own limitation, its CPU DAI driver should set the supported format by its own circumstance, since the soc-pcm driver will use the intersection of cpu_dai's formats and codec_dai's formats. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_receiver.c | 2 ++ sound/soc/codecs/spdif_transmitter.c | 5 +++-- 2 files changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index e9d7881ed2c8..26d34744c677 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -25,6 +25,8 @@ #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) static struct snd_soc_codec_driver soc_codec_spdif_dir; diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index 18280499fd55..efc3d88d7f8c 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -25,8 +25,9 @@ #define DRV_NAME "spdif-dit" #define STUB_RATES SNDRV_PCM_RATE_8000_96000 -#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE - +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_codec_driver soc_codec_spdif_dit; -- cgit v1.2.3-70-g09d2 From 4f8b19143d74e1c3360b21640065765a12bafb1b Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 31 Jul 2013 13:28:52 +0100 Subject: ASoC: wm0010: Fix resource leak If kzalloc() fails for `img' then we are going to leak the memory for `out'. We are freeing the memory of all the tx/rx transfers but the tx/rx buf pointers will be NULL if we drop out earlier. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f5e835662cdc..10adc4145d46 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -410,6 +410,16 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) rec->command, rec->length); len = rec->length + 8; + xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); + if (!xfer) { + dev_err(codec->dev, "Failed to allocate xfer\n"); + ret = -ENOMEM; + goto abort; + } + + xfer->codec = codec; + list_add_tail(&xfer->list, &xfer_list); + out = kzalloc(len, GFP_KERNEL); if (!out) { dev_err(codec->dev, @@ -417,6 +427,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.rx_buf = out; img = kzalloc(len, GFP_KERNEL); if (!img) { @@ -425,24 +436,13 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.tx_buf = img; byte_swap_64((u64 *)&rec->command, img, len); - xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); - if (!xfer) { - dev_err(codec->dev, "Failed to allocate xfer\n"); - ret = -ENOMEM; - goto abort1; - } - - xfer->codec = codec; - list_add_tail(&xfer->list, &xfer_list); - spi_message_init(&xfer->m); xfer->m.complete = wm0010_boot_xfer_complete; xfer->m.context = xfer; - xfer->t.tx_buf = img; - xfer->t.rx_buf = out; xfer->t.len = len; xfer->t.bits_per_word = 8; -- cgit v1.2.3-70-g09d2 From 3f1a91aa25579ba5e7268a47a73d2a83e4802c62 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 29 Jul 2013 18:37:32 -0300 Subject: ASoC: fsl: Fix module build Building imx_v6_v7_defconfig with all audio drivers as modules results in the folowing build error: ERROR: "imx_pcm_fiq_init" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_dma_init" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_fiq_exit" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_dma_exit" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_dma_init" [sound/soc/fsl/snd-soc-fsl-ssi.ko] undefined! ERROR: "imx_pcm_dma_exit" [sound/soc/fsl/snd-soc-fsl-ssi.ko] undefined! Fix this by allowing SND_SOC_IMX_PCM_FIQ and SND_SOC_IMX_PCM_DMA to be also built as modules and by using 'IS_ENABLED' to cover the module case. Reported-by: Guennadi Liakhovetski Signed-off-by: Fabio Estevam Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 4 ++-- sound/soc/fsl/imx-pcm.h | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3a79d01e8d03..c26449b54270 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -109,11 +109,11 @@ config SND_SOC_IMX_SSI tristate config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SOC_IMX_AUDMUX diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 9136625a3460..5d5b73303e11 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -38,7 +38,7 @@ struct imx_pcm_fiq_params { struct snd_dmaengine_dai_dma_data *dma_params_tx; }; -#ifdef CONFIG_SND_SOC_IMX_PCM_DMA +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) int imx_pcm_dma_init(struct platform_device *pdev); void imx_pcm_dma_exit(struct platform_device *pdev); #else @@ -52,7 +52,7 @@ static inline void imx_pcm_dma_exit(struct platform_device *pdev) } #endif -#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ) int imx_pcm_fiq_init(struct platform_device *pdev, struct imx_pcm_fiq_params *params); void imx_pcm_fiq_exit(struct platform_device *pdev); -- cgit v1.2.3-70-g09d2 From 70263cb474853c116f80713d468f3c17d805921c Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 30 Jul 2013 07:51:37 +0800 Subject: ASoC: rcar: fix return value check in rsnd_gen1_probe() In case of error, the function devm_ioremap_resource() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(), and also remove the dev_err call to avoid redundant error message. Signed-off-by: Wei Yongjun Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 17 ++++------------- 1 file changed, 4 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 5e4ae0da4352..61232cd9908f 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -150,25 +150,16 @@ static int rsnd_gen1_probe(struct platform_device *pdev, sru_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SRU); adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_ADG); ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SSI); - if (!sru_res || - !adg_res || - !ssi_res) { - dev_err(dev, "Not enough SRU/SSI/ADG platform resources.\n"); - return -ENODEV; - } - - gen->ops = &rsnd_gen1_ops; gen->base[RSND_GEN1_SRU] = devm_ioremap_resource(dev, sru_res); gen->base[RSND_GEN1_ADG] = devm_ioremap_resource(dev, adg_res); gen->base[RSND_GEN1_SSI] = devm_ioremap_resource(dev, ssi_res); - if (!gen->base[RSND_GEN1_SRU] || - !gen->base[RSND_GEN1_ADG] || - !gen->base[RSND_GEN1_SSI]) { - dev_err(dev, "SRU/SSI/ADG ioremap failed\n"); + if (IS_ERR(gen->base[RSND_GEN1_SRU]) || + IS_ERR(gen->base[RSND_GEN1_ADG]) || + IS_ERR(gen->base[RSND_GEN1_SSI])) return -ENODEV; - } + gen->ops = &rsnd_gen1_ops; rsnd_gen1_reg_map_init(gen); dev_dbg(dev, "Gen1 device probed\n"); -- cgit v1.2.3-70-g09d2 From 8fe120b5a665fc869c23f86e4964b801f6e53486 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 31 Jul 2013 19:00:32 +0200 Subject: ASoC: omap-abe-twl6040: Remove support for pdata (legacy boot) Just recently OMAP4 legacy boot support has been removed. No reason to keep the code used by the legacy boot (pdata based) since neither OMAP4 or OMAP5 can boot in this mode. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- include/linux/platform_data/omap-abe-twl6040.h | 49 --------- sound/soc/omap/omap-abe-twl6040.c | 133 ++++++++----------------- 2 files changed, 41 insertions(+), 141 deletions(-) delete mode 100644 include/linux/platform_data/omap-abe-twl6040.h (limited to 'sound') diff --git a/include/linux/platform_data/omap-abe-twl6040.h b/include/linux/platform_data/omap-abe-twl6040.h deleted file mode 100644 index 5d298ac10fc2..000000000000 --- a/include/linux/platform_data/omap-abe-twl6040.h +++ /dev/null @@ -1,49 +0,0 @@ -/** - * omap-abe-twl6040.h - ASoC machine driver OMAP4+ devices, header. - * - * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com - * All rights reserved. - * - * Author: Peter Ujfalusi - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef _OMAP_ABE_TWL6040_H_ -#define _OMAP_ABE_TWL6040_H_ - -/* To select if only one channel is connected in a stereo port */ -#define ABE_TWL6040_LEFT (1 << 0) -#define ABE_TWL6040_RIGHT (1 << 1) - -struct omap_abe_twl6040_data { - char *card_name; - /* Feature flags for connected audio pins */ - u8 has_hs; - u8 has_hf; - bool has_ep; - u8 has_aux; - u8 has_vibra; - bool has_dmic; - bool has_hsmic; - bool has_mainmic; - bool has_submic; - u8 has_afm; - /* Other features */ - bool jack_detection; /* board can detect jack events */ - int mclk_freq; /* MCLK frequency speed for twl6040 */ -}; - -#endif /* _OMAP_ABE_TWL6040_H_ */ diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 70cd5c7b2e14..ebb13906b3a0 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -23,7 +23,6 @@ #include #include #include -#include #include #include @@ -166,19 +165,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFMR", NULL, "Line In"}, }; -static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm, - int connected, char *pin) -{ - if (!connected) - snd_soc_dapm_disable_pin(dapm, pin); -} - static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; - struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int hs_trim; int ret = 0; @@ -203,24 +193,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - /* - * NULL pdata means we booted with DT. In this case the routing is - * provided and the card is fully routed, no need to mark pins. - */ - if (!pdata) - return ret; - - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - return ret; } @@ -274,13 +246,18 @@ static struct snd_soc_card omap_abe_card = { static int omap_abe_probe(struct platform_device *pdev) { - struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; + struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; int ret = 0; + if (!node) { + dev_err(&pdev->dev, "of node is missing.\n"); + return -ENODEV; + } + card->dev = &pdev->dev; priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); @@ -289,78 +266,50 @@ static int omap_abe_probe(struct platform_device *pdev) priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (node) { - struct device_node *dai_node; - - if (snd_soc_of_parse_card_name(card, "ti,model")) { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } - ret = snd_soc_of_parse_audio_routing(card, - "ti,audio-routing"); - if (ret) { - dev_err(&pdev->dev, - "Error while parsing DAPM routing\n"); - return ret; - } + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "Error while parsing DAPM routing\n"); + return ret; + } - dai_node = of_parse_phandle(node, "ti,mcpdm", 0); - if (!dai_node) { - dev_err(&pdev->dev, "McPDM node is not provided\n"); - return -EINVAL; - } - abe_twl6040_dai_links[0].cpu_dai_name = NULL; - abe_twl6040_dai_links[0].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_of_node = dai_node; - dai_node = of_parse_phandle(node, "ti,dmic", 0); - if (dai_node) { - num_links = 2; - abe_twl6040_dai_links[1].cpu_dai_name = NULL; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_of_node = dai_node; - priv->dmic_codec_dev = platform_device_register_simple( + priv->dmic_codec_dev = platform_device_register_simple( "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, - "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } - } else { - num_links = 1; - } - - priv->jack_detection = of_property_read_bool(node, - "ti,jack-detection"); - of_property_read_u32(node, "ti,mclk-freq", - &priv->mclk_freq); - if (!priv->mclk_freq) { - dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); } - - omap_abe_card.fully_routed = 1; - } else if (pdata) { - if (pdata->card_name) { - card->name = pdata->card_name; - } else { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } - - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; + num_links = 1; + } + + priv->jack_detection = of_property_read_bool(node, "ti,jack-detection"); + of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; } + card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); -- cgit v1.2.3-70-g09d2 From d4780eec779c4e6d2fe5963dd2aee0a85d956122 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 1 Aug 2013 09:53:45 +0100 Subject: ASoC: wm0010: Use DMA-safe memory for SPI transfers We should be allocating our buffers for the SPI transfers from the DMA zone. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 10adc4145d46..d5ebcb00019b 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -420,7 +420,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) xfer->codec = codec; list_add_tail(&xfer->list, &xfer_list); - out = kzalloc(len, GFP_KERNEL); + out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate RX buffer\n"); @@ -429,7 +429,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) } xfer->t.rx_buf = out; - img = kzalloc(len, GFP_KERNEL); + img = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img) { dev_err(codec->dev, "Failed to allocate image buffer\n"); @@ -523,14 +523,14 @@ static int wm0010_stage2_load(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ - img = kzalloc(fw->size, GFP_KERNEL); + img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!img) { dev_err(codec->dev, "Failed to allocate image buffer\n"); ret = -ENOMEM; goto abort2; } - out = kzalloc(fw->size, GFP_KERNEL); + out = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate output buffer\n"); ret = -ENOMEM; @@ -670,14 +670,14 @@ static int wm0010_boot(struct snd_soc_codec *codec) ret = -ENOMEM; len = pll_rec.length + 8; - out = kzalloc(len, GFP_KERNEL); + out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate RX buffer\n"); goto abort; } - img_swap = kzalloc(len, GFP_KERNEL); + img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img_swap) { dev_err(codec->dev, "Failed to allocate image buffer\n"); -- cgit v1.2.3-70-g09d2 From 95169d080fcaad6c990ce3602d9b3d38753b1fa4 Mon Sep 17 00:00:00 2001 From: Marek Belisko Date: Thu, 1 Aug 2013 11:14:58 +0200 Subject: ASoC: Add PCM1681 codec driver. PCM1681 can be controlled via I2C, SPI or in bootstrap mode (no control mode). This code add support only for I2C mode. Signed-off-by: Marek Belisko Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1681.c | 313 +++++++++++++++++++++ 4 files changed, 334 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 sound/soc/codecs/pcm1681.c (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt new file mode 100644 index 000000000000..4df17185ab80 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt @@ -0,0 +1,15 @@ +Texas Instruments PCM1681 8-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,pcm1681". + - reg: The i2c address. Should contain <0x4c>. + +Examples: + + i2c_bus { + pcm1681@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..e2daf820179b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1681 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -292,6 +293,9 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1681 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd8066f546..4a068d20444d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -42,6 +42,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c new file mode 100644 index 000000000000..27da41b2dfcd --- /dev/null +++ b/sound/soc/codecs/pcm1681.c @@ -0,0 +1,313 @@ +/* + * PCM1681 ASoC codec driver + * + * Copyright (c) StreamUnlimited GmbH 2013 + * Marek Belisko + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1681_SOFT_MUTE_ALL 0xff +#define PCM1681_DEEMPH_RATE_MASK 0x18 +#define PCM1681_DEEMPH_MASK 0x01 + +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ + +static const struct reg_default pcm1681_reg_defaults[] = { + { 0x01, 0xff }, + { 0x02, 0xff }, + { 0x03, 0xff }, + { 0x04, 0xff }, + { 0x05, 0xff }, + { 0x06, 0xff }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x06 }, + { 0x0A, 0x00 }, + { 0x0B, 0xff }, + { 0x0C, 0x0f }, + { 0x0D, 0x00 }, + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, +}; + +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x00) || (reg == 0x0f)); +} + +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) +{ + return pcm1681_accessible_reg(dev, reg) && + (reg != PCM1681_ZERO_DETECT_STATUS); +} + +struct pcm1681_private { + struct regmap *regmap; + unsigned int format; + /* Current deemphasis status */ + unsigned int deemph; + /* Current rate for deemphasis control */ + unsigned int rate; +}; + +static const int pcm1681_deemph[] = { 44100, 48000, 32000 }; + +static int pcm1681_set_deemph(struct snd_soc_codec *codec) +{ + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int i = 0, val = -1, enable = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) + if (pcm1681_deemph[i] == priv->rate) + val = i; + + if (val != -1) { + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_RATE_MASK, val); + enable = 1; + } else + enable = 0; + + /* enable/disable deemphasis functionality */ + return regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_MASK, enable); +} + +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return pcm1681_set_deemph(codec); +} + +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The PCM1681 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + priv->format = format; + + return 0; +} + +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val; + + if (mute) + val = PCM1681_SOFT_MUTE_ALL; + else + val = 0; + + return regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); +} + +static int pcm1681_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) + val = 0x00; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x03; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x04; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x05; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); + if (ret < 0) + return ret; + + return pcm1681_set_deemph(codec); +} + +static const struct snd_soc_dai_ops pcm1681_dai_ops = { + .set_fmt = pcm1681_set_dai_fmt, + .hw_params = pcm1681_hw_params, + .digital_mute = pcm1681_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); + +static const struct snd_kcontrol_new pcm1681_controls[] = { + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 7/8 Playback Volume", + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + pcm1681_get_deemph, pcm1681_put_deemph), +}; + +struct snd_soc_dai_driver pcm1681_dai = { + .name = "pcm1681-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = PCM1681_PCM_RATES, + .formats = PCM1681_PCM_FORMATS, + }, + .ops = &pcm1681_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1681_dt_ids[] = { + { .compatible = "ti,pcm1681", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); +#endif + +static const struct regmap_config pcm1681_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(pcm1681_reg_defaults) + 1, + .reg_defaults = pcm1681_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), + .writeable_reg = pcm1681_writeable_reg, + .readable_reg = pcm1681_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { + .controls = pcm1681_controls, + .num_controls = ARRAY_SIZE(pcm1681_controls), +}; + +static const struct i2c_device_id pcm1681_i2c_id[] = { + {"pcm1681", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); + +static int pcm1681_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + int ret; + struct pcm1681_private *priv; + + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(client, priv); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681, + &pcm1681_dai, 1); +} + +static int pcm1681_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver pcm1681_i2c_driver = { + .driver = { + .name = "pcm1681", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pcm1681_dt_ids), + }, + .id_table = pcm1681_i2c_id, + .probe = pcm1681_i2c_probe, + .remove = pcm1681_i2c_remove, +}; + +module_i2c_driver(pcm1681_i2c_driver); + +MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Marek Belisko "); +MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From 92bb4c32708ee3e1d6eb0e185d678dab35152daf Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 1 Aug 2013 11:11:28 +0100 Subject: ASoC: wm_adsp: Sanitize parameter passing No need to hold on to the `codec' pointer. We can use the `dsp' pointer and grab all the information we need from there. This makes the parameters for the functions a bit more sane and idiomatic. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 31 ++++++++++++++++--------------- sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 17 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3168224bc104..b38f3506418f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -227,7 +227,6 @@ struct wm_coeff_ctl_ops { struct wm_coeff_ctl { const char *name; - struct snd_soc_card *card; struct wm_adsp_alg_region region; struct wm_coeff_ctl_ops ops; struct wm_adsp *adsp; @@ -484,7 +483,7 @@ static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) struct snd_kcontrol_new *kcontrol; int ret; - if (!ctl || !ctl->name || !ctl->card) + if (!ctl || !ctl->name) return -EINVAL; kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); @@ -498,14 +497,14 @@ static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) kcontrol->put = wm_coeff_put; kcontrol->private_value = (unsigned long)ctl; - ret = snd_soc_add_card_controls(ctl->card, + ret = snd_soc_add_card_controls(adsp->card, kcontrol, 1); if (ret < 0) goto err_kcontrol; kfree(kcontrol); - ctl->kcontrol = snd_soc_card_get_kcontrol(ctl->card, + ctl->kcontrol = snd_soc_card_get_kcontrol(adsp->card, ctl->name); list_add(&ctl->list, &adsp->ctl_list); @@ -777,11 +776,10 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(ctl_work); } -static int wm_adsp_create_control(struct snd_soc_codec *codec, +static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *region) { - struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); struct wm_coeff_ctl *ctl; struct wmfw_ctl_work *ctl_work; char *name; @@ -840,7 +838,6 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, ctl->set = 0; ctl->ops.xget = wm_coeff_get; ctl->ops.xput = wm_coeff_put; - ctl->card = codec->card; ctl->adsp = dsp; ctl->len = region->len; @@ -877,7 +874,7 @@ err_name: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) +static int wm_adsp_setup_algs(struct wm_adsp *dsp) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -1065,7 +1062,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].dm); region->len -= be32_to_cpu(adsp1_alg[i].dm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region DM with ID %x\n", be32_to_cpu(adsp1_alg[i].alg.id)); @@ -1082,7 +1079,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].zm); region->len -= be32_to_cpu(adsp1_alg[i].zm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", be32_to_cpu(adsp1_alg[i].alg.id)); @@ -1111,7 +1108,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].xm); region->len -= be32_to_cpu(adsp2_alg[i].xm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region XM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1128,7 +1125,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].ym); region->len -= be32_to_cpu(adsp2_alg[i].ym); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region YM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1145,7 +1142,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].zm); region->len -= be32_to_cpu(adsp2_alg[i].zm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1365,6 +1362,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, int ret; int val; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, @@ -1399,7 +1398,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp, codec); + ret = wm_adsp_setup_algs(dsp); if (ret != 0) goto err; @@ -1492,6 +1491,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, unsigned int val; int ret; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: /* @@ -1554,7 +1555,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp, codec); + ret = wm_adsp_setup_algs(dsp); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 64087fb1cdec..d018dea6254d 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -39,6 +39,7 @@ struct wm_adsp { int type; struct device *dev; struct regmap *regmap; + struct snd_soc_card *card; int base; int sysclk_reg; -- cgit v1.2.3-70-g09d2 From 2f6f0ffb2b073a0a5a9ffe5705b8e8cc43558d3a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 1 Aug 2013 11:02:47 +0100 Subject: ASoC: samsung: Make secondary I2S DAI device a child of primary More for neatness than for any great utility. Really we shouldn't be creating the child device at all, refactoring will follow. Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 973735841a05..849ac0e225ca 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1019,6 +1019,8 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) if (IS_ERR(i2s->pdev)) return NULL; + i2s->pdev->dev.parent = &pdev->dev; + platform_set_drvdata(i2s->pdev, i2s); ret = platform_device_add(i2s->pdev); if (ret < 0) -- cgit v1.2.3-70-g09d2 From 9356e9d51c80114fce2d7d8be99bce1d7e19d063 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Aug 2013 14:08:06 +0200 Subject: ASoC: dapm: Check return value of snd_soc_cnew() snd_soc_cnew() can return NULL, so we should check the result before trying to use it. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d74c3560d556..b4fae8717851 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -671,8 +671,10 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], NULL, name, prefix); - kcontrol->private_free = dapm_kcontrol_free; kfree(long_name); + if (!kcontrol) + return -ENOMEM; + kcontrol->private_free = dapm_kcontrol_free; ret = dapm_kcontrol_data_alloc(w, kcontrol); if (ret) { -- cgit v1.2.3-70-g09d2 From 868ead653e7f65a9ac05777d0736a181a3c1c150 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 1 Aug 2013 19:29:22 +0800 Subject: ASoC: rt5640: remove unused mux Remove unused "INL Mux" and "INR Mux". Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 26 -------------------------- 1 file changed, 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index ce585e37e38a..4db7314baabc 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -737,29 +737,6 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; -/* INL/R source */ -static const char * const rt5640_inl_src[] = { - "IN2P", "MONOP" -}; - -static const SOC_ENUM_SINGLE_DECL( - rt5640_inl_enum, RT5640_INL_INR_VOL, - RT5640_INL_SEL_SFT, rt5640_inl_src); - -static const struct snd_kcontrol_new rt5640_inl_mux = - SOC_DAPM_ENUM("INL source", rt5640_inl_enum); - -static const char * const rt5640_inr_src[] = { - "IN2N", "MONON" -}; - -static const SOC_ENUM_SINGLE_DECL( - rt5640_inr_enum, RT5640_INL_INR_VOL, - RT5640_INR_SEL_SFT, rt5640_inr_src); - -static const struct snd_kcontrol_new rt5640_inr_mux = - SOC_DAPM_ENUM("INR source", rt5640_inr_enum); - /* Stereo ADC source */ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" @@ -1005,9 +982,6 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { RT5640_PWR_IN_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR VOL", RT5640_PWR_VOL, RT5640_PWR_IN_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL Mux", SND_SOC_NOPM, 0, 0, &rt5640_inl_mux), - SND_SOC_DAPM_MUX("INR Mux", SND_SOC_NOPM, 0, 0, &rt5640_inr_mux), /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5640_PWR_MIXER, RT5640_PWR_RM_L_BIT, 0, rt5640_rec_l_mix, ARRAY_SIZE(rt5640_rec_l_mix)), -- cgit v1.2.3-70-g09d2 From f091f3f07328f75d20a2a5970d1f8b58d95fc990 Mon Sep 17 00:00:00 2001 From: Lothar Waßmann Date: Wed, 31 Jul 2013 16:44:29 +0200 Subject: ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When a sound capture/playback is terminated while a playback/capture is running, power_vag_event() will clear SGTL5000_CHIP_ANA_POWER in the SND_SOC_DAPM_PRE_PMD event, thus muting the respective other channel. Don't clear SGTL5000_CHIP_ANA_POWER when both DAC and ADC are active to prevent this. Signed-off-by: Lothar Waßmann Reviewed-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6c8a9e7bee25..9303c7d011b2 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -153,6 +153,8 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP; + switch (event) { case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, @@ -160,9 +162,17 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); - msleep(400); + /* + * Don't clear VAG_POWERUP, when both DAC and ADC are + * operational to prevent inadvertently starving the + * other one of them. + */ + if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) & + mask) != mask) { + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + } break; default: break; -- cgit v1.2.3-70-g09d2 From 65f2b226763bc348a9b9145aa5e17e7e3f6d8c35 Mon Sep 17 00:00:00 2001 From: Lothar Waßmann Date: Wed, 31 Jul 2013 16:44:30 +0200 Subject: ASoC: sgtl5000: fix buggy 'Capture Attenuate Switch' control MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The SGTL5000 Capture Attenuate Switch (or "ADC Volume Range Reduction" as it is called in the manual) is single bit only. Signed-off-by: Lothar Waßmann Reviewed-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9303c7d011b2..760e8bfeacaa 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -398,7 +398,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0), SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)", SGTL5000_CHIP_ANA_ADC_CTRL, - 8, 2, 0, capture_6db_attenuate), + 8, 1, 0, capture_6db_attenuate), SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), SOC_DOUBLE_TLV("Headphone Playback Volume", -- cgit v1.2.3-70-g09d2 From 2c75bdf3fd935119cf8681ac0df2b4a5edd5167d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Aug 2013 14:08:07 +0200 Subject: ASoC: dapm: Fix kcontrol path list corruption When calling krealloc for the kcontrol data the items in the path list that point back to the head of the list will now point to freed memory, which causes the list to become corrupted. To fix this, instead of resizing the whole data struct, only resize the widget list. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 38 +++++++++++++++++++++----------------- 1 file changed, 21 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b4fae8717851..5f64c16336ad 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -177,7 +177,7 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( struct dapm_kcontrol_data { unsigned int value; struct list_head paths; - struct snd_soc_dapm_widget_list wlist; + struct snd_soc_dapm_widget_list *wlist; }; static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, @@ -185,7 +185,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, { struct dapm_kcontrol_data *data; - data = kzalloc(sizeof(*data) + sizeof(widget), GFP_KERNEL); + data = kzalloc(sizeof(*data), GFP_KERNEL); if (!data) { dev_err(widget->dapm->dev, "ASoC: can't allocate kcontrol data for %s\n", @@ -193,8 +193,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, return -ENOMEM; } - data->wlist.widgets[0] = widget; - data->wlist.num_widgets = 1; INIT_LIST_HEAD(&data->paths); kcontrol->private_data = data; @@ -205,6 +203,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); + kfree(data->wlist); kfree(data); } @@ -213,25 +212,30 @@ static struct snd_soc_dapm_widget_list *dapm_kcontrol_get_wlist( { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); - return &data->wlist; + return data->wlist; } static int dapm_kcontrol_add_widget(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget *widget) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); - struct dapm_kcontrol_data *new_data; - unsigned int n = data->wlist.num_widgets + 1; + struct snd_soc_dapm_widget_list *new_wlist; + unsigned int n; + + if (data->wlist) + n = data->wlist->num_widgets + 1; + else + n = 1; - new_data = krealloc(data, sizeof(*data) + sizeof(widget) * n, - GFP_KERNEL); - if (!new_data) + new_wlist = krealloc(data->wlist, + sizeof(*new_wlist) + sizeof(widget) * n, GFP_KERNEL); + if (!new_wlist) return -ENOMEM; - new_data->wlist.widgets[n - 1] = widget; - new_data->wlist.num_widgets = n; + new_wlist->widgets[n - 1] = widget; + new_wlist->num_widgets = n; - kcontrol->private_data = new_data; + data->wlist = new_wlist; return 0; } @@ -689,12 +693,12 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, w->name, name, ret); return ret; } - } else { - ret = dapm_kcontrol_add_widget(kcontrol, w); - if (ret) - return ret; } + ret = dapm_kcontrol_add_widget(kcontrol, w); + if (ret) + return ret; + w->kcontrols[kci] = kcontrol; dapm_kcontrol_add_path(kcontrol, path); -- cgit v1.2.3-70-g09d2 From fe581391147cb3d738d961d0f1233d91a9e1113c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Aug 2013 18:30:38 +0200 Subject: ASoC: dapm: Fix empty list check in dapm_new_mux() list_first_entry() will always return a valid pointer, even if the list is empty. So the check whether path is NULL will always be false. So we end up calling dapm_create_or_share_mixmux_kcontrol() with a path struct that points right in the middle of the widget struct and by trying to modify the path the widgets memory will become corrupted. Fix this by using list_emtpy() to check if the widget doesn't have any paths. Signed-off-by: Lars-Peter Clausen Tested-by: Stephen Warren Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-dapm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bd16010441cc..4375c9f2b791 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -679,13 +679,14 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - path = list_first_entry(&w->sources, struct snd_soc_dapm_path, - list_sink); - if (!path) { + if (list_empty(&w->sources)) { dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name); return -EINVAL; } + path = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From 697aebab78a88c6b164cfb74d19b86817d2ccd82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Aug 2013 08:38:27 +0200 Subject: ALSA: hda - Fix missing fixup for Mac Mini with STAC9221 A fixup for Apple Mac Mini was lost during the adaption to the generic parser because the fallback for the generic ID 8384:7680 was dropped, and it resulted in the silence output (and maybe other problems). Unfortunately, just adding the missing subsystem ID wasn't enough, in this case. The subsystem ID of this machine is 0000:0100 (what Apple thought...?), and since snd_hda_pick_fixup() doesn't take the vendor id zero into account, the driver ignored this entry. Now it's fixed to regard the vendor id zero as a valid value. Reported-and-tested-by: Linus Torvalds Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/patch_sigmatel.c | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 7c11d46b84d3..48a9d004d6d9 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -860,7 +860,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } if (id < 0 && quirk) { - for (q = quirk; q->subvendor; q++) { + for (q = quirk; q->subvendor || q->subdevice; q++) { unsigned int vendorid = q->subdevice | (q->subvendor << 16); unsigned int mask = 0xffff0000 | q->subdevice_mask; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 92b9b4324372..6d1924c19abf 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2819,6 +2819,7 @@ static const struct hda_pintbl ecs202_pin_configs[] = { /* codec SSIDs for Intel Mac sharing the same PCI SSID 8384:7680 */ static const struct snd_pci_quirk stac922x_intel_mac_fixup_tbl[] = { + SND_PCI_QUIRK(0x0000, 0x0100, "Mac Mini", STAC_INTEL_MAC_V3), SND_PCI_QUIRK(0x106b, 0x0800, "Mac", STAC_INTEL_MAC_V1), SND_PCI_QUIRK(0x106b, 0x0600, "Mac", STAC_INTEL_MAC_V2), SND_PCI_QUIRK(0x106b, 0x0700, "Mac", STAC_INTEL_MAC_V2), -- cgit v1.2.3-70-g09d2 From 6ad709482e151068b7197f4572edeeae5eeaff93 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 4 Aug 2013 20:32:04 +0100 Subject: ASoC: spdif_transceiver: add output pin widget CODECs without any outputs now remain powered down, which means any paths to these codecs also remain powered down. Add an always-enabled output pin widget to the spdif transceiver codec. This enables DAPM to correctly identify that the spdif transceiver is in use when playback is enabled, which will then allow DAPM to power up any links from the CPU DAI to the S/PDIF transceiver. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_transmitter.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index efc3d88d7f8c..4e96d10d61a2 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -29,7 +29,20 @@ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_codec_driver soc_codec_spdif_dit; +static const struct snd_soc_dapm_widget dit_widgets[] = { + SND_SOC_DAPM_OUTPUT("spdif-out"), +}; + +static const const struct snd_soc_dapm_route dit_routes[] = { + { "spdif-out", NULL, "Playback" }, +}; + +static struct snd_soc_codec_driver soc_codec_spdif_dit = { + .dapm_widgets = dit_widgets, + .num_dapm_widgets = ARRAY_SIZE(dit_widgets), + .dapm_routes = dit_routes, + .num_dapm_routes = ARRAY_SIZE(dit_routes), +}; static struct snd_soc_dai_driver dit_stub_dai = { .name = "dit-hifi", -- cgit v1.2.3-70-g09d2 From a7d094297946e32da9bdf03cd5be1f6954d17ed3 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 4 Aug 2013 20:22:03 +0100 Subject: ASoC: kirkwood: merge struct kirkwood_dma_priv with struct kirkwood_dma_data Merge these two structures together; nothing other than the I2S and DMA driver makes use of struct kirkwood_dma_data, and it's not like struct kirkwood_dma_data is really just used to convey DMA specific data to the backend; it's more a general shared structure between the two halves. This will later allow kirkwood-dma.c and kirkwood-i2s.c to be merged together. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 76 ++++++++++++--------------------------- sound/soc/kirkwood/kirkwood.h | 2 ++ 2 files changed, 24 insertions(+), 54 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index a9f14530c3db..ba50dd156c67 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -33,11 +33,11 @@ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE) -struct kirkwood_dma_priv { - struct snd_pcm_substream *play_stream; - struct snd_pcm_substream *rec_stream; - struct kirkwood_dma_data *data; -}; +static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) +{ + struct snd_soc_pcm_runtime *soc_runtime = subs->private_data; + return snd_soc_dai_get_drvdata(soc_runtime->cpu_dai); +} static struct snd_pcm_hardware kirkwood_dma_snd_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | @@ -63,8 +63,7 @@ static u64 kirkwood_dma_dmamask = DMA_BIT_MASK(32); static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) { - struct kirkwood_dma_priv *prdata = dev_id; - struct kirkwood_dma_data *priv = prdata->data; + struct kirkwood_dma_data *priv = dev_id; unsigned long mask, status, cause; mask = readl(priv->io + KIRKWOOD_INT_MASK); @@ -89,10 +88,10 @@ static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) writel(status, priv->io + KIRKWOOD_INT_CAUSE); if (status & KIRKWOOD_INT_CAUSE_PLAY_BYTES) - snd_pcm_period_elapsed(prdata->play_stream); + snd_pcm_period_elapsed(priv->substream_play); if (status & KIRKWOOD_INT_CAUSE_REC_BYTES) - snd_pcm_period_elapsed(prdata->rec_stream); + snd_pcm_period_elapsed(priv->substream_rec); return IRQ_HANDLED; } @@ -126,15 +125,10 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) { int err; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_platform *platform = soc_runtime->platform; - struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; - struct kirkwood_dma_data *priv; - struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform); + struct kirkwood_dma_data *priv = kirkwood_priv(substream); const struct mbus_dram_target_info *dram; unsigned long addr; - priv = snd_soc_dai_get_dma_data(cpu_dai, substream); snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw); /* Ensure that all constraints linked to dma burst are fulfilled */ @@ -157,21 +151,11 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) if (err < 0) return err; - if (prdata == NULL) { - prdata = kzalloc(sizeof(struct kirkwood_dma_priv), GFP_KERNEL); - if (prdata == NULL) - return -ENOMEM; - - prdata->data = priv; - + if (!priv->substream_play && !priv->substream_rec) { err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, - "kirkwood-i2s", prdata); - if (err) { - kfree(prdata); + "kirkwood-i2s", priv); + if (err) return -EBUSY; - } - - snd_soc_platform_set_drvdata(platform, prdata); /* * Enable Error interrupts. We're only ack'ing them but @@ -183,11 +167,11 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) dram = mv_mbus_dram_info(); addr = substream->dma_buffer.addr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - prdata->play_stream = substream; + priv->substream_play = substream; kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_PLAYBACK_WIN, addr, dram); } else { - prdata->rec_stream = substream; + priv->substream_rec = substream; kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_RECORD_WIN, addr, dram); } @@ -197,27 +181,19 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) static int kirkwood_dma_close(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; - struct snd_soc_platform *platform = soc_runtime->platform; - struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform); - struct kirkwood_dma_data *priv; - - priv = snd_soc_dai_get_dma_data(cpu_dai, substream); + struct kirkwood_dma_data *priv = kirkwood_priv(substream); - if (!prdata || !priv) + if (!priv) return 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - prdata->play_stream = NULL; + priv->substream_play = NULL; else - prdata->rec_stream = NULL; + priv->substream_rec = NULL; - if (!prdata->play_stream && !prdata->rec_stream) { + if (!priv->substream_play && !priv->substream_rec) { writel(0, priv->io + KIRKWOOD_ERR_MASK); - free_irq(priv->irq, prdata); - kfree(prdata); - snd_soc_platform_set_drvdata(platform, NULL); + free_irq(priv->irq, priv); } return 0; @@ -243,13 +219,9 @@ static int kirkwood_dma_hw_free(struct snd_pcm_substream *substream) static int kirkwood_dma_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; - struct kirkwood_dma_data *priv; + struct kirkwood_dma_data *priv = kirkwood_priv(substream); unsigned long size, count; - priv = snd_soc_dai_get_dma_data(cpu_dai, substream); - /* compute buffer size in term of "words" as requested in specs */ size = frames_to_bytes(runtime, runtime->buffer_size); size = (size>>2)-1; @@ -272,13 +244,9 @@ static int kirkwood_dma_prepare(struct snd_pcm_substream *substream) static snd_pcm_uframes_t kirkwood_dma_pointer(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; - struct kirkwood_dma_data *priv; + struct kirkwood_dma_data *priv = kirkwood_priv(substream); snd_pcm_uframes_t count; - priv = snd_soc_dai_get_dma_data(cpu_dai, substream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) count = bytes_to_frames(substream->runtime, readl(priv->io + KIRKWOOD_PLAY_BYTE_COUNT)); diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 4d92637ddb3f..10a3aaafe0aa 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -129,6 +129,8 @@ struct kirkwood_dma_data { struct clk *extclk; uint32_t ctl_play; uint32_t ctl_rec; + struct snd_pcm_substream *substream_play; + struct snd_pcm_substream *substream_rec; int irq; int burst; }; -- cgit v1.2.3-70-g09d2 From 57295073b6acfdfaf9319d3caf92a5c433fdf109 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 5 Aug 2013 11:27:31 +0200 Subject: ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 17 +++++++- include/sound/soc.h | 28 ++++++------ sound/soc/soc-dapm.c | 108 +++++++++++++++++++++++++++++++++++++++++------ 3 files changed, 125 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3575721a955d..c728d28ae9a5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -280,14 +280,26 @@ struct device; { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_AUTODISABLE(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_AUTODISABLE(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ @@ -484,6 +496,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_in, /* link to DAI structure */ snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ + snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ }; enum snd_soc_dapm_subclass { diff --git a/include/sound/soc.h b/include/sound/soc.h index b1e1f967ae1e..6201c6ede8ba 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -30,13 +30,13 @@ /* * Convenience kcontrol builders */ -#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert) \ +#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert, xautodisable) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, .shift = shift_left, \ .rshift = shift_right, .max = xmax, .platform_max = xmax, \ - .invert = xinvert}) -#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \ - SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert) + .invert = xinvert, .autodisable = xautodisable}) +#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \ + SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert}) @@ -52,7 +52,7 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ @@ -68,7 +68,7 @@ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -97,7 +97,7 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \ - max, invert) } + max, invert, 0) } #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw, \ @@ -119,7 +119,7 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \ - max, invert) } + max, invert, 0) } #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -190,14 +190,14 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ - .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) } #define SOC_DOUBLE_EXT(xname, reg, shift_left, shift_right, max, invert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = \ - SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert) } + SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert, 0) } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -206,7 +206,7 @@ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ - .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) } #define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -216,7 +216,7 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, \ - xmax, xinvert) } + xmax, xinvert, 0) } #define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -1088,7 +1088,9 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift, invert; + unsigned int reg, rreg, shift, rshift; + unsigned int invert:1; + unsigned int autodisable:1; }; struct soc_bytes { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5f64c16336ad..0944bc4bd4a4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -47,6 +47,15 @@ #define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++; +static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink, + const char *control, + int (*connected)(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink)); +static struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget); + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -73,16 +82,18 @@ static int dapm_up_seq[] = { [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, [snd_soc_dapm_line] = 10, - [snd_soc_dapm_post] = 11, + [snd_soc_dapm_kcontrol] = 11, + [snd_soc_dapm_post] = 12, }; static int dapm_down_seq[] = { [snd_soc_dapm_pre] = 0, - [snd_soc_dapm_adc] = 1, - [snd_soc_dapm_hp] = 2, - [snd_soc_dapm_spk] = 2, - [snd_soc_dapm_line] = 2, - [snd_soc_dapm_out_drv] = 2, + [snd_soc_dapm_kcontrol] = 1, + [snd_soc_dapm_adc] = 2, + [snd_soc_dapm_hp] = 3, + [snd_soc_dapm_spk] = 3, + [snd_soc_dapm_line] = 3, + [snd_soc_dapm_out_drv] = 3, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, @@ -176,6 +187,7 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( struct dapm_kcontrol_data { unsigned int value; + struct snd_soc_dapm_widget *widget; struct list_head paths; struct snd_soc_dapm_widget_list *wlist; }; @@ -184,6 +196,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data; + struct soc_mixer_control *mc; data = kzalloc(sizeof(*data), GFP_KERNEL); if (!data) { @@ -195,6 +208,39 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, INIT_LIST_HEAD(&data->paths); + switch (widget->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + mc = (struct soc_mixer_control *)kcontrol->private_value; + + if (mc->autodisable) { + struct snd_soc_dapm_widget template; + + memset(&template, 0, sizeof(template)); + template.reg = mc->reg; + template.mask = (1 << fls(mc->max)) - 1; + template.shift = mc->shift; + if (mc->invert) + template.off_val = mc->max; + else + template.off_val = 0; + template.on_val = template.off_val; + template.id = snd_soc_dapm_kcontrol; + template.name = kcontrol->id.name; + + data->widget = snd_soc_dapm_new_control(widget->dapm, + &template); + if (!data->widget) { + kfree(data); + return -ENOMEM; + } + } + break; + default: + break; + } + kcontrol->private_data = data; return 0; @@ -203,6 +249,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); + kfree(data->widget); kfree(data->wlist); kfree(data); } @@ -246,6 +293,21 @@ static void dapm_kcontrol_add_path(const struct snd_kcontrol *kcontrol, struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); list_add_tail(&path->list_kcontrol, &data->paths); + + if (data->widget) { + snd_soc_dapm_add_path(data->widget->dapm, data->widget, + path->source, NULL, NULL); + } +} + +static bool dapm_kcontrol_is_powered(const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + if (!data->widget) + return true; + + return data->widget->power; } static struct list_head *dapm_kcontrol_get_path_list( @@ -275,6 +337,9 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, if (data->value == value) return false; + if (data->widget) + data->widget->on_val = value; + data->value = value; return true; @@ -515,6 +580,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_spk: case snd_soc_dapm_line: case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: p->connect = 1; break; /* does affect routing - dynamically connected */ @@ -880,6 +946,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: return 0; default: break; @@ -975,6 +1042,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: return 0; default: break; @@ -1523,7 +1591,7 @@ static void dapm_widget_update(struct snd_soc_card *card) unsigned int wi; int ret; - if (!update) + if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) return; wlist = dapm_kcontrol_get_wlist(update->kcontrol); @@ -1668,6 +1736,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -2335,6 +2404,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -2717,6 +2787,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -2724,17 +2795,24 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + unsigned int val; if (snd_soc_volsw_is_stereo(mc)) dev_warn(codec->dapm.dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); - ucontrol->value.integer.value[0] = - (snd_soc_read(codec, reg) >> shift) & mask; + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + if (dapm_kcontrol_is_powered(kcontrol)) + val = (snd_soc_read(codec, reg) >> shift) & mask; + else + val = dapm_kcontrol_get_value(kcontrol); + mutex_unlock(&card->dapm_mutex); + if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; + ucontrol->value.integer.value[0] = max - val; + else + ucontrol->value.integer.value[0] = val; return 0; } @@ -2775,11 +2853,14 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (invert) val = max - val; - mask = mask << shift; - val = val << shift; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + dapm_kcontrol_set_value(kcontrol, val); + + mask = mask << shift; + val = val << shift; + change = snd_soc_test_bits(codec, reg, mask, val); if (change) { update.kcontrol = kcontrol; @@ -3179,6 +3260,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: w->power_check = dapm_supply_check_power; break; default: -- cgit v1.2.3-70-g09d2 From e06e4c2d530fd4995c41083009647263ccd77d3b Mon Sep 17 00:00:00 2001 From: Oskar Schirmer Date: Mon, 5 Aug 2013 07:36:02 +0000 Subject: ASoC: sgtl5000: fix codec clock source transition to avoid clockless moment Powering down PLL before switching to a mode that does not use it is a bad idea. It would cause the SGTL5000 be without internal clock supply, especially on the I2C interface, which would make subsequent access to it fail. Thus, in case of not using PLL any longer, first set the mode control, then power down PLL. Signed-off-by: Oskar Schirmer Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7c99f3ccb1c6..54ca169ec27e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -644,16 +644,19 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); } else { + /* otherwise, clk_ctrl must be set before pll power down */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + /* power down pll */ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, 0); } - /* if using pll, clk_ctrl must be set after pll power up */ - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); - return 0; } -- cgit v1.2.3-70-g09d2 From 9d58a077465ff23b935042bf1cbdac64cdb78a2c Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 5 Aug 2013 13:17:28 +0100 Subject: ASoC: core: init delayed_work for codec-codec links We must init the delayed_work for codec-codec links otherwise shutting down the DAI chain will fault when calling flush_delayed_work_sync() on the linked DAI. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0ec070cf7231..2940e2c04525 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -530,6 +530,15 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif +static void codec2codec_close_delayed_work(struct work_struct *work) +{ + /* Currently nothing to do for c2c links + * Since c2c links are internal nodes in the DAPM graph and + * don't interface with the outside world or application layer + * we don't have to do any special handling on close. + */ +} + #ifdef CONFIG_PM_SLEEP /* powers down audio subsystem for suspend */ int snd_soc_suspend(struct device *dev) @@ -1428,6 +1437,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) return ret; } } else { + INIT_DELAYED_WORK(&rtd->delayed_work, + codec2codec_close_delayed_work); + /* link the DAI widgets */ play_w = codec_dai->playback_widget; capture_w = cpu_dai->capture_widget; -- cgit v1.2.3-70-g09d2 From af64d7341ab51335eeb03453180cf200b120ec43 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 4 Aug 2013 20:23:03 +0100 Subject: ASoC: kirkwood: Free external clock if it is a duplicate of internal [Remaining patch from "ASoC: kirkwood: use devm_clk_get() for the external clock" -- broonie] Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index ba7203995369..0109b1e8449a 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -495,6 +495,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { + devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); } else { dev_info(&pdev->dev, "found external clock\n"); -- cgit v1.2.3-70-g09d2 From 19c2c5f55e31ac8da87bb8efe0cf86aa933e6a2f Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 4 Aug 2013 20:24:03 +0100 Subject: ASoC: avoid duplicated DAI routes ASoC automatically creates snd_soc_dapm_dai_in and snd_soc_dapm_dai_out widgets for DAI drivers, and adds them to the list. Later on, ASoC creates automatic routes between these widgets and a widget with a stream name. We look for a snd_soc_dapm_dai_in or snd_soc_dapm_dai_out widget, and use this to obtain the DAI structure. We then scan all widgets for any with a stream name refering to either the capture or the playback stream, and create routes. If you have both a snd_soc_dapm_dai_in and a snd_soc_dapm_dai_out referring to the same DAI structure, this ends up creating one set of routes for the DAI for the snd_soc_dapm_dai_in widget, and a duplicated set of routes for the snd_soc_dapm_dai_out widget. Fix this by checking that the stream name for the widget matches the DAI widget name. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0944bc4bd4a4..7f53d8662297 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3551,7 +3551,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) break; } - if (!w->sname) + if (!w->sname || !strstr(w->sname, dai_w->name)) continue; if (dai->driver->playback.stream_name && -- cgit v1.2.3-70-g09d2 From 13b02fa0dbb1311d08dfacd897a6ff41232d7cfb Mon Sep 17 00:00:00 2001 From: Michael Trimarchi Date: Sat, 3 Aug 2013 16:20:43 +0200 Subject: ASoC: Add PCM1792A spi mode codec support Add PCM1792A spi mode codec support. This version implements only a subset of functionalities. Tested connect to a pandaboard ES device and based on recently pcm1681 codec. Signed-off-by: Michael Trimarchi Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/pcm1792a.txt | 18 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1792a.c | 245 +++++++++++++++++++++ sound/soc/codecs/pcm1792a.h | 26 +++ 5 files changed, 295 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm1792a.txt create mode 100644 sound/soc/codecs/pcm1792a.c create mode 100644 sound/soc/codecs/pcm1792a.h (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/pcm1792a.txt b/Documentation/devicetree/bindings/sound/pcm1792a.txt new file mode 100644 index 000000000000..970ba1ed576f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm1792a.txt @@ -0,0 +1,18 @@ +Texas Instruments pcm1792a DT bindings + +This driver supports the SPI bus. + +Required properties: + + - compatible: "ti,pcm1792a" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Examples: + + codec_spi: 1792a@0 { + compatible = "ti,pcm1792a"; + spi-max-frequency = <600000>; + }; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..4afe94330820 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -292,6 +293,9 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1792A + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd8066f546..811ca12febcb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -42,6 +42,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c new file mode 100644 index 000000000000..3f83bf9895d4 --- /dev/null +++ b/sound/soc/codecs/pcm1792a.c @@ -0,0 +1,245 @@ +/* + * PCM1792A ASoC codec driver + * + * Copyright (c) Amarula Solutions B.V. 2013 + * + * Michael Trimarchi + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include + +#include "pcm1792a.h" + +#define PCM1792A_DAC_VOL_LEFT 0x10 +#define PCM1792A_DAC_VOL_RIGHT 0x11 +#define PCM1792A_FMT_CONTROL 0x12 +#define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL + +#define PCM1792A_FMT_MASK 0x70 +#define PCM1792A_FMT_SHIFT 4 +#define PCM1792A_MUTE_MASK 0x01 +#define PCM1792A_MUTE_SHIFT 0 +#define PCM1792A_ATLD_ENABLE (1 << 7) + +static const struct reg_default pcm1792a_reg_defaults[] = { + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x50 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x01 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, +}; + +static bool pcm1792a_accessible_reg(struct device *dev, unsigned int reg) +{ + return reg >= 0x10 && reg <= 0x17; +} + +static bool pcm1792a_writeable_reg(struct device *dev, unsigned register reg) +{ + bool accessible; + + accessible = pcm1792a_accessible_reg(dev, reg); + + return accessible && reg != 0x16 && reg != 0x17; +} + +struct pcm1792a_private { + struct regmap *regmap; + unsigned int format; + unsigned int rate; +}; + +static int pcm1792a_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->format = format; + + return 0; +} + +static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, + PCM1792A_MUTE_MASK, !!mute); + if (ret < 0) + return ret; + + return 0; +} + +static int pcm1792a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x02; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x05; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x04; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE; + + ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, + PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val); + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_soc_dai_ops pcm1792a_dai_ops = { + .set_fmt = pcm1792a_set_dai_fmt, + .hw_params = pcm1792a_hw_params, + .digital_mute = pcm1792a_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1792a_dac_tlv, -12000, 50, 1); + +static const struct snd_kcontrol_new pcm1792a_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT, + PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0, + pcm1792a_dac_tlv), +}; + +static struct snd_soc_dai_driver pcm1792a_dai = { + .name = "pcm1792a-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PCM1792A_RATES, + .formats = PCM1792A_FORMATS, }, + .capture = { + .channels_min = 0, + .channels_max = 0, + }, + .ops = &pcm1792a_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1792a_of_match[] = { + { .compatible = "ti,pcm1792a", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1792a_of_match); +#endif + +static const struct regmap_config pcm1792a_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 24, + .reg_defaults = pcm1792a_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1792a_reg_defaults), + .writeable_reg = pcm1792a_writeable_reg, + .readable_reg = pcm1792a_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1792a = { + .controls = pcm1792a_controls, + .num_controls = ARRAY_SIZE(pcm1792a_controls), +}; + +static int pcm1792a_spi_probe(struct spi_device *spi) +{ + struct pcm1792a_private *pcm1792a; + int ret; + + pcm1792a = devm_kzalloc(&spi->dev, sizeof(struct pcm1792a_private), + GFP_KERNEL); + if (!pcm1792a) + return -ENOMEM; + + spi_set_drvdata(spi, pcm1792a); + + pcm1792a->regmap = devm_regmap_init_spi(spi, &pcm1792a_regmap); + if (IS_ERR(pcm1792a->regmap)) { + ret = PTR_ERR(pcm1792a->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&spi->dev, + &soc_codec_dev_pcm1792a, &pcm1792a_dai, 1); +} + +static int pcm1792a_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm1792a_spi_ids[] = { + { "pcm1792a", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm1792a_spi_ids); + +static struct spi_driver pcm1792a_codec_driver = { + .driver = { + .name = "pcm1792a", + .owner = THIS_MODULE, + .of_match_table = pcm1792a_of_match, + }, + .id_table = pcm1792a_spi_ids, + .probe = pcm1792a_spi_probe, + .remove = pcm1792a_spi_remove, +}; + +module_spi_driver(pcm1792a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM1792A driver"); +MODULE_AUTHOR("Michael Trimarchi "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h new file mode 100644 index 000000000000..7a83d1fc102a --- /dev/null +++ b/sound/soc/codecs/pcm1792a.h @@ -0,0 +1,26 @@ +/* + * definitions for PCM1792A + * + * Copyright 2013 Amarula Solutions + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __PCM1792A_H__ +#define __PCM1792A_H__ + +#define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S16_LE) + +#endif -- cgit v1.2.3-70-g09d2 From db43b16fa0e913582b63c971848e08151d50d952 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 4 Aug 2013 20:26:03 +0100 Subject: ASoC: kirkwood: provide KIRKWOOD_PLAYCTL_ENABLE_MASK Provide a helper macro which includes the sum of all enable bits in the playback control register. This simplifies the code a little. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 10 ++++------ sound/soc/kirkwood/kirkwood.h | 5 ++++- 2 files changed, 8 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0109b1e8449a..ad1c789637b2 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -198,8 +198,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, ctl_play |= KIRKWOOD_PLAYCTL_MONO_OFF; priv->ctl_play &= ~(KIRKWOOD_PLAYCTL_MONO_MASK | - KIRKWOOD_PLAYCTL_I2S_EN | - KIRKWOOD_PLAYCTL_SPDIF_EN | + KIRKWOOD_PLAYCTL_ENABLE_MASK | KIRKWOOD_PLAYCTL_SIZE_MASK); priv->ctl_play |= ctl_play; } else { @@ -243,8 +242,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_play; - value = ctl & ~(KIRKWOOD_PLAYCTL_I2S_EN | - KIRKWOOD_PLAYCTL_SPDIF_EN); + value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); /* enable interrupts */ @@ -266,7 +264,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, writel(value, priv->io + KIRKWOOD_INT_MASK); /* disable all playbacks */ - ctl &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN); + ctl &= ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -386,7 +384,7 @@ static int kirkwood_i2s_probe(struct snd_soc_dai *dai) /* disable playback/record */ value = readl(priv->io + KIRKWOOD_PLAYCTL); - value &= ~(KIRKWOOD_PLAYCTL_I2S_EN|KIRKWOOD_PLAYCTL_SPDIF_EN); + value &= ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_RECCTL); diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 10a3aaafe0aa..9a50607267cf 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -54,7 +54,7 @@ #define KIRKWOOD_PLAYCTL_MONO_OFF (0<<5) #define KIRKWOOD_PLAYCTL_I2S_MUTE (1<<7) #define KIRKWOOD_PLAYCTL_SPDIF_EN (1<<4) -#define KIRKWOOD_PLAYCTL_I2S_EN (1<<3) +#define KIRKWOOD_PLAYCTL_I2S_EN (1<<3) #define KIRKWOOD_PLAYCTL_SIZE_MASK (7<<0) #define KIRKWOOD_PLAYCTL_SIZE_16 (7<<0) #define KIRKWOOD_PLAYCTL_SIZE_16_C (3<<0) @@ -62,6 +62,9 @@ #define KIRKWOOD_PLAYCTL_SIZE_24 (1<<0) #define KIRKWOOD_PLAYCTL_SIZE_32 (0<<0) +#define KIRKWOOD_PLAYCTL_ENABLE_MASK (KIRKWOOD_PLAYCTL_SPDIF_EN | \ + KIRKWOOD_PLAYCTL_I2S_EN) + #define KIRKWOOD_PLAY_BUF_ADDR 0x1104 #define KIRKWOOD_PLAY_BUF_SIZE 0x1108 #define KIRKWOOD_PLAY_BYTE_COUNT 0x110C -- cgit v1.2.3-70-g09d2 From 64ddf1f89cd7a483e1204320395023774234b49a Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 4 Aug 2013 20:27:03 +0100 Subject: ASoC: kirkwood: combine kirkwood-i2s and kirkwood-dma drivers These really should be a single driver because they're fully integrated in hardware. Make them so. Signed-off-by: Russell King Signed-off-by: Mark Brown --- arch/arm/mach-dove/common.c | 4 ++-- arch/arm/mach-kirkwood/common.c | 24 +++++++++--------------- sound/soc/kirkwood/Kconfig | 5 ----- sound/soc/kirkwood/Makefile | 4 +--- sound/soc/kirkwood/kirkwood-dma.c | 30 +----------------------------- sound/soc/kirkwood/kirkwood-i2s.c | 21 ++++++++++++++++----- sound/soc/kirkwood/kirkwood-openrd.c | 4 ++-- sound/soc/kirkwood/kirkwood-t5325.c | 4 ++-- sound/soc/kirkwood/kirkwood.h | 2 ++ 9 files changed, 35 insertions(+), 63 deletions(-) (limited to 'sound') diff --git a/arch/arm/mach-dove/common.c b/arch/arm/mach-dove/common.c index 00247c771313..304f069ebf50 100644 --- a/arch/arm/mach-dove/common.c +++ b/arch/arm/mach-dove/common.c @@ -108,8 +108,8 @@ static void __init dove_clk_init(void) orion_clkdev_add(NULL, "sdhci-dove.1", sdio1); orion_clkdev_add(NULL, "orion_nand", nand); orion_clkdev_add(NULL, "cafe1000-ccic.0", camera); - orion_clkdev_add(NULL, "kirkwood-i2s.0", i2s0); - orion_clkdev_add(NULL, "kirkwood-i2s.1", i2s1); + orion_clkdev_add(NULL, "mvebu-audio.0", i2s0); + orion_clkdev_add(NULL, "mvebu-audio.1", i2s1); orion_clkdev_add(NULL, "mv_crypto", crypto); orion_clkdev_add(NULL, "dove-ac97", ac97); orion_clkdev_add(NULL, "dove-pdma", pdma); diff --git a/arch/arm/mach-kirkwood/common.c b/arch/arm/mach-kirkwood/common.c index e9238b5567ee..1663de090984 100644 --- a/arch/arm/mach-kirkwood/common.c +++ b/arch/arm/mach-kirkwood/common.c @@ -264,7 +264,7 @@ void __init kirkwood_clk_init(void) orion_clkdev_add(NULL, MV_XOR_NAME ".1", xor1); orion_clkdev_add("0", "pcie", pex0); orion_clkdev_add("1", "pcie", pex1); - orion_clkdev_add(NULL, "kirkwood-i2s", audio); + orion_clkdev_add(NULL, "mvebu-audio", audio); orion_clkdev_add(NULL, MV64XXX_I2C_CTLR_NAME ".0", runit); orion_clkdev_add(NULL, MV64XXX_I2C_CTLR_NAME ".1", runit); @@ -560,7 +560,7 @@ void __init kirkwood_timer_init(void) /***************************************************************************** * Audio ****************************************************************************/ -static struct resource kirkwood_i2s_resources[] = { +static struct resource kirkwood_audio_resources[] = { [0] = { .start = AUDIO_PHYS_BASE, .end = AUDIO_PHYS_BASE + SZ_16K - 1, @@ -573,29 +573,23 @@ static struct resource kirkwood_i2s_resources[] = { }, }; -static struct kirkwood_asoc_platform_data kirkwood_i2s_data = { +static struct kirkwood_asoc_platform_data kirkwood_audio_data = { .burst = 128, }; -static struct platform_device kirkwood_i2s_device = { - .name = "kirkwood-i2s", +static struct platform_device kirkwood_audio_device = { + .name = "mvebu-audio", .id = -1, - .num_resources = ARRAY_SIZE(kirkwood_i2s_resources), - .resource = kirkwood_i2s_resources, + .num_resources = ARRAY_SIZE(kirkwood_audio_resources), + .resource = kirkwood_audio_resources, .dev = { - .platform_data = &kirkwood_i2s_data, + .platform_data = &kirkwood_audio_data, }, }; -static struct platform_device kirkwood_pcm_device = { - .name = "kirkwood-pcm-audio", - .id = -1, -}; - void __init kirkwood_audio_init(void) { - platform_device_register(&kirkwood_i2s_device); - platform_device_register(&kirkwood_pcm_device); + platform_device_register(&kirkwood_audio_device); } /***************************************************************************** diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 59085ad6c41a..9e1970c44e86 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -6,14 +6,10 @@ config SND_KIRKWOOD_SOC the Kirkwood I2S interface. You will also need to select the audio interfaces to support below. -config SND_KIRKWOOD_SOC_I2S - tristate - config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) depends on I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help Say Y if you want to add support for SoC audio on @@ -22,7 +18,6 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_ALC5623 help Say Y if you want to add support for SoC audio on diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 3e62ae9e7bbe..9e781385cb88 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -1,8 +1,6 @@ -snd-soc-kirkwood-objs := kirkwood-dma.o -snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o +snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o snd-soc-openrd-objs := kirkwood-openrd.o snd-soc-t5325-objs := kirkwood-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index ba50dd156c67..01622f6358df 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -334,36 +334,8 @@ static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm) } } -static struct snd_soc_platform_driver kirkwood_soc_platform = { +struct snd_soc_platform_driver kirkwood_soc_platform = { .ops = &kirkwood_dma_ops, .pcm_new = kirkwood_dma_new, .pcm_free = kirkwood_dma_free_dma_buffers, }; - -static int kirkwood_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform); -} - -static int kirkwood_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver kirkwood_pcm_driver = { - .driver = { - .name = "kirkwood-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = kirkwood_soc_platform_probe, - .remove = kirkwood_soc_platform_remove, -}; - -module_platform_driver(kirkwood_pcm_driver); - -MODULE_AUTHOR("Arnaud Patard "); -MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:kirkwood-pcm-audio"); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index ad1c789637b2..e5f3f7a9ea26 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -24,7 +24,7 @@ #include #include "kirkwood.h" -#define DRV_NAME "kirkwood-i2s" +#define DRV_NAME "mvebu-audio" #define KIRKWOOD_I2S_FORMATS \ (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -517,10 +517,20 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, soc_dai, 1); - if (!err) - return 0; - dev_err(&pdev->dev, "snd_soc_register_component failed\n"); + if (err) { + dev_err(&pdev->dev, "snd_soc_register_component failed\n"); + goto err_component; + } + err = snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform); + if (err) { + dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); + goto err_platform; + } + return 0; + err_platform: + snd_soc_unregister_component(&pdev->dev); + err_component: if (!IS_ERR(priv->extclk)) clk_disable_unprepare(priv->extclk); clk_disable_unprepare(priv->clk); @@ -532,6 +542,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) { struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_component(&pdev->dev); if (!IS_ERR(priv->extclk)) @@ -556,4 +567,4 @@ module_platform_driver(kirkwood_i2s_driver); MODULE_AUTHOR("Arnaud Patard, "); MODULE_DESCRIPTION("Kirkwood I2S SoC Interface"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:kirkwood-i2s"); +MODULE_ALIAS("platform:mvebu-audio"); diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index addbebc2b3fa..025be0e97164 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -52,8 +52,8 @@ static struct snd_soc_dai_link openrd_client_dai[] = { { .name = "CS42L51", .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .cpu_dai_name = "mvebu-audio", + .platform_name = "mvebu-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 4f4cb56f765a..27545b0c4856 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -68,8 +68,8 @@ static struct snd_soc_dai_link t5325_dai[] = { { .name = "ALC5621", .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .cpu_dai_name = "mvebu-audio", + .platform_name = "mvebu-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 9a50607267cf..1d13dee93d90 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -138,4 +138,6 @@ struct kirkwood_dma_data { int burst; }; +extern struct snd_soc_platform_driver kirkwood_soc_platform; + #endif -- cgit v1.2.3-70-g09d2 From e4065f3ff122e35cfc760d9a712564f3d9ef3a49 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 4 Aug 2013 20:28:04 +0100 Subject: ASoC: kirkwood: move calculation of max buffer size to kirkwood.h Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/kirkwood/kirkwood.h | 2 ++ 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 01622f6358df..b238434f92b0 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -51,7 +51,7 @@ static struct snd_pcm_hardware kirkwood_dma_snd_hw = { .rate_max = 384000, .channels_min = 1, .channels_max = 8, - .buffer_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES * KIRKWOOD_SND_MAX_PERIODS, + .buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES, .period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES, .period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES, .periods_min = KIRKWOOD_SND_MIN_PERIODS, diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 1d13dee93d90..f8e1ccc1c58c 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -125,6 +125,8 @@ #define KIRKWOOD_SND_MAX_PERIODS 16 #define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x4000 #define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x4000 +#define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \ + * KIRKWOOD_SND_MAX_PERIODS) struct kirkwood_dma_data { void __iomem *io; -- cgit v1.2.3-70-g09d2 From 55af2d23c6c9caf7da6c9a55bbea83dccbc1af2b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 5 Aug 2013 18:20:53 +0100 Subject: ASoC: pcm1792a: Fix build with !OF Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 3f83bf9895d4..72cf8353e812 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -169,13 +169,11 @@ static struct snd_soc_dai_driver pcm1792a_dai = { .ops = &pcm1792a_dai_ops, }; -#ifdef CONFIG_OF static const struct of_device_id pcm1792a_of_match[] = { { .compatible = "ti,pcm1792a", }, { } }; MODULE_DEVICE_TABLE(of, pcm1792a_of_match); -#endif static const struct regmap_config pcm1792a_regmap = { .reg_bits = 8, @@ -231,7 +229,7 @@ static struct spi_driver pcm1792a_codec_driver = { .driver = { .name = "pcm1792a", .owner = THIS_MODULE, - .of_match_table = pcm1792a_of_match, + .of_match_table = of_match_ptr(pcm1792a_of_match), }, .id_table = pcm1792a_spi_ids, .probe = pcm1792a_spi_probe, -- cgit v1.2.3-70-g09d2 From d66a5b9c82f2a2a6d424a7ccad51c52f150fa181 Mon Sep 17 00:00:00 2001 From: Lothar Waßmann Date: Fri, 2 Aug 2013 10:30:15 +0200 Subject: ASoC: mxs: add some error messages to help identifying problems MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Lothar Waßmann Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 29 +++++++++++++++++++++++------ 1 file changed, 23 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index b2e372dd02eb..ce084eb10c49 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -50,18 +50,27 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, } /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */ - if (mclk < 8000000 || mclk > 27000000) + if (mclk < 8000000 || mclk > 27000000) { + dev_err(codec_dai->dev, "Invalid mclk frequency: %u.%03uMHz\n", + mclk / 1000000, mclk / 1000 % 1000); return -EINVAL; + } /* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */ ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0); - if (ret) + if (ret) { + dev_err(codec_dai->dev, "Failed to set sysclk to %u.%03uMHz\n", + mclk / 1000000, mclk / 1000 % 1000); return ret; + } /* The SAIF MCLK should be the same as SGTL5000_SYSCLK */ ret = snd_soc_dai_set_sysclk(cpu_dai, MXS_SAIF_MCLK, mclk, 0); - if (ret) + if (ret) { + dev_err(cpu_dai->dev, "Failed to set sysclk to %u.%03uMHz\n", + mclk / 1000000, mclk / 1000 % 1000); return ret; + } /* set codec to slave mode */ dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | @@ -69,13 +78,19 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, dai_format); - if (ret) + if (ret) { + dev_err(codec_dai->dev, "Failed to set dai format to %08x\n", + dai_format); return ret; + } /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); - if (ret) + if (ret) { + dev_err(cpu_dai->dev, "Failed to set dai format to %08x\n", + dai_format); return ret; + } return 0; } @@ -153,8 +168,10 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev) * should be >= 8MHz and <= 27M. */ ret = mxs_saif_get_mclk(0, 44100 * 256, 44100); - if (ret) + if (ret) { + dev_err(&pdev->dev, "failed to get mclk\n"); return ret; + } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- cgit v1.2.3-70-g09d2 From ed6a27723979cfffab62c450baba4f75ebcbda78 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 5 Aug 2013 23:34:17 +0100 Subject: ASoC: wm8994: Fix class W controls Commit 6e0650 (ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding it) went too far and converted a DAPM control to use SOC_SINGLE_EXT() which crashes. Revert that portion of the patch. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 24131a7f9390..c99b6da79efd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1433,7 +1433,7 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, #define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ - snd_soc_get_volsw, wm8994_put_class_w) + snd_soc_dapm_get_volsw, wm8994_put_class_w) static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.2.3-70-g09d2 From f8f11795b96a3632edb25a8924c61bfb74581cb0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 6 Aug 2013 13:39:29 +0200 Subject: ASoC: tlv320aic26: Fix keyclick feature The tlv320aic26 contains a embedded snd_soc_codec struct which is referenced in the keyclick code. That struct is never initialized though, replace the embedded struct with a pointer and use that in the keyclick code. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b1f6982c7c9c..b192cd4705a0 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -29,7 +29,7 @@ MODULE_LICENSE("GPL"); /* AIC26 driver private data */ struct aic26 { struct spi_device *spi; - struct snd_soc_codec codec; + struct snd_soc_codec *codec; int master; int datfm; int mclk; @@ -330,7 +330,7 @@ static ssize_t aic26_keyclick_show(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; - val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); amp = (val >> 12) & 0x7; freq = (125 << ((val >> 8) & 0x7)) >> 1; len = 2 * (1 + ((val >> 4) & 0xf)); @@ -346,9 +346,9 @@ static ssize_t aic26_keyclick_set(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val; - val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); val |= 0x8000; - aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + aic26_reg_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); return count; } @@ -360,8 +360,11 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); */ static int aic26_probe(struct snd_soc_codec *codec) { + struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, err, i, reg; + aic26->codec = codec; + dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); /* Reset the codec to power on defaults */ -- cgit v1.2.3-70-g09d2 From 95ad868289a24dbc072412ce2fb0d40cb34c5794 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 6 Aug 2013 13:39:31 +0200 Subject: ASoC: mc13783: Remove embedded snd_soc_codec structs from private data structs It is unused and a leftover of the pre multi-component era. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 5402dfbbb716..4d3c8fd8c5db 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -94,7 +94,6 @@ #define AUDIO_DAC_CFS_DLY_B (1 << 10) struct mc13783_priv { - struct snd_soc_codec codec; struct mc13xxx *mc13xxx; enum mc13783_ssi_port adc_ssi_port; -- cgit v1.2.3-70-g09d2 From 0d59ff3a24ad099f741da5efd9e3e02bfd64496e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 6 Aug 2013 13:39:30 +0200 Subject: ASoC: twl4030: Remove embedded snd_soc_codec structs from private data structs It is unused and a leftover of the pre multi-component era. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 8e6e5b016021..1e3884d6b3fb 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -137,8 +137,6 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { /* codec private data */ struct twl4030_priv { - struct snd_soc_codec codec; - unsigned int codec_powered; /* reference counts of AIF/APLL users */ -- cgit v1.2.3-70-g09d2 From c7f3843575eac1eea1fbda2f6b61d36627fa8f7c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 6 Aug 2013 17:03:55 +0100 Subject: ASoC: wm5110: Correct input OSR bits for wm5110 The input OSR bits are specified differently for wm5110 than for current revs of wm5102. This patch corrects support for this on wm5110. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 20 ++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 12 ++++-------- 3 files changed, 25 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8dc6881496de..779a0eeac67c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -553,6 +553,26 @@ const struct soc_enum arizona_ng_hold = 4, arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); +static const char * const arizona_in_dmic_osr_text[] = { + "1.536MHz", "3.072MHz", "6.144MHz", +}; + +const struct soc_enum arizona_in_dmic_osr[] = { + SOC_ENUM_SINGLE(ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN2L_CONTROL, ARIZONA_IN2_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), +}; +EXPORT_SYMBOL_GPL(arizona_in_dmic_osr); + static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index fe1b794bd5f0..b6b6d7036ea0 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -198,6 +198,7 @@ extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; extern const struct soc_enum arizona_ng_hold; +extern const struct soc_enum arizona_in_dmic_osr[]; extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fc410377277f..77fd531bf3cc 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -58,14 +58,10 @@ static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); SOC_SINGLE(name " NG SPKDAT2R Switch", base, 11, 1, 0) static const struct snd_kcontrol_new wm5110_snd_controls[] = { -SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, - ARIZONA_IN4_OSR_SHIFT, 1, 0), +SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]), +SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), +SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]), +SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]), SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -- cgit v1.2.3-70-g09d2 From 4b4dab82340d969521f4f86108441cb597c8595d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 28 Jul 2013 18:58:29 -0700 Subject: ASoC: rsnd: remove platform dai and add dai_id on platform setting Current rsnd driver is using struct rsnd_dai_platform_info so that indicate sound DAI information (playback/capture SSI ID). But, SSI settings were also required separately. Thus, platform settings was very un-understandable. This patch adds dai_id to SSI settings, and removed rsnd_dai_platform_info. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 18 ++++++++------- sound/soc/sh/rcar/core.c | 60 +++++++++++++++++++++++++++++++++--------------- sound/soc/sh/rcar/gen.c | 10 ++++---- sound/soc/sh/rcar/rsnd.h | 3 +++ sound/soc/sh/rcar/ssi.c | 22 ++++++++++++++++++ 5 files changed, 82 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index 99d8dd029906..33233edd1664 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -28,15 +28,24 @@ /* * flags * - * 0xA0000000 + * 0xAB000000 * * A : clock sharing settings + * B : SSI direction */ #define RSND_SSI_CLK_PIN_SHARE (1 << 31) #define RSND_SSI_CLK_FROM_ADG (1 << 30) /* clock parent is master */ #define RSND_SSI_SYNC (1 << 29) /* SSI34_sync etc */ +#define RSND_SSI_PLAY (1 << 24) + +#define RSND_SSI_SET(_dai_id, _pio_irq, _flags) \ +{ .dai_id = _dai_id, .pio_irq = _pio_irq, .flags = _flags } +#define RSND_SSI_UNUSED \ +{ .dai_id = -1, .pio_irq = -1, .flags = 0 } + struct rsnd_ssi_platform_info { + int dai_id; int pio_irq; u32 flags; }; @@ -45,11 +54,6 @@ struct rsnd_scu_platform_info { u32 flags; }; -struct rsnd_dai_platform_info { - int ssi_id_playback; - int ssi_id_capture; -}; - /* * flags * @@ -66,8 +70,6 @@ struct rcar_snd_info { int ssi_info_nr; struct rsnd_scu_platform_info *scu_info; int scu_info_nr; - struct rsnd_dai_platform_info *dai_info; - int dai_info_nr; int (*start)(int id); int (*stop)(int id); }; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 9a5469d3f352..420d6df9c3d0 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -219,6 +219,16 @@ int rsnd_dai_disconnect(struct rsnd_mod *mod) return 0; } +int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) +{ + int id = rdai - priv->rdai; + + if ((id < 0) || (id >= rsnd_dai_nr(priv))) + return -EINVAL; + + return id; +} + struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id) { return priv->rdai + id; @@ -315,9 +325,10 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct rsnd_priv *priv = snd_soc_dai_get_drvdata(dai); struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); - struct rsnd_dai_platform_info *info = rsnd_dai_get_platform_info(rdai); - int ssi_id = rsnd_dai_is_play(rdai, io) ? info->ssi_id_playback : - info->ssi_id_capture; + struct rsnd_mod *mod = rsnd_ssi_mod_get_frm_dai(priv, + rsnd_dai_id(priv, rdai), + rsnd_dai_is_play(rdai, io)); + int ssi_id = rsnd_mod_id(mod); int ret; unsigned long flags; @@ -439,10 +450,24 @@ static int rsnd_dai_probe(struct platform_device *pdev, { struct snd_soc_dai_driver *drv; struct rsnd_dai *rdai; + struct rsnd_mod *pmod, *cmod; struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_dai_platform_info *dai_info; - int dai_nr = info->dai_info_nr; - int i, pid, cid; + int dai_nr; + int i; + + /* get max dai nr */ + for (dai_nr = 0; dai_nr < 32; dai_nr++) { + pmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 1); + cmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 0); + + if (!pmod && !cmod) + break; + } + + if (!dai_nr) { + dev_err(dev, "no dai\n"); + return -EIO; + } drv = devm_kzalloc(dev, sizeof(*drv) * dai_nr, GFP_KERNEL); rdai = devm_kzalloc(dev, sizeof(*rdai) * dai_nr, GFP_KERNEL); @@ -452,10 +477,9 @@ static int rsnd_dai_probe(struct platform_device *pdev, } for (i = 0; i < dai_nr; i++) { - dai_info = &info->dai_info[i]; - pid = dai_info->ssi_id_playback; - cid = dai_info->ssi_id_capture; + pmod = rsnd_ssi_mod_get_frm_dai(priv, i, 1); + cmod = rsnd_ssi_mod_get_frm_dai(priv, i, 0); /* * init rsnd_dai @@ -463,8 +487,6 @@ static int rsnd_dai_probe(struct platform_device *pdev, INIT_LIST_HEAD(&rdai[i].playback.head); INIT_LIST_HEAD(&rdai[i].capture.head); - rdai[i].info = dai_info; - snprintf(rdai[i].name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", i); /* @@ -472,20 +494,22 @@ static int rsnd_dai_probe(struct platform_device *pdev, */ drv[i].name = rdai[i].name; drv[i].ops = &rsnd_soc_dai_ops; - if (pid >= 0) { + if (pmod) { drv[i].playback.rates = RSND_RATES; drv[i].playback.formats = RSND_FMTS; drv[i].playback.channels_min = 2; drv[i].playback.channels_max = 2; } - if (cid >= 0) { + if (cmod) { drv[i].capture.rates = RSND_RATES; drv[i].capture.formats = RSND_FMTS; drv[i].capture.channels_min = 2; drv[i].capture.channels_max = 2; } - dev_dbg(dev, "%s (%d, %d) probed", rdai[i].name, pid, cid); + dev_dbg(dev, "%s (%s/%s)\n", rdai[i].name, + pmod ? "play" : " -- ", + cmod ? "capture" : " -- "); } priv->dai_nr = dai_nr; @@ -627,10 +651,6 @@ static int rsnd_probe(struct platform_device *pdev) if (ret < 0) return ret; - ret = rsnd_dai_probe(pdev, info, priv); - if (ret < 0) - return ret; - ret = rsnd_scu_probe(pdev, info, priv); if (ret < 0) return ret; @@ -643,6 +663,10 @@ static int rsnd_probe(struct platform_device *pdev) if (ret < 0) return ret; + ret = rsnd_dai_probe(pdev, info, priv); + if (ret < 0) + return ret; + /* * asoc register */ diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 61232cd9908f..460c57eef267 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -49,7 +49,6 @@ static int rsnd_gen1_path_init(struct rsnd_priv *priv, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) { - struct rsnd_dai_platform_info *info = rsnd_dai_get_platform_info(rdai); struct rsnd_mod *mod; int ret; int id; @@ -67,10 +66,11 @@ static int rsnd_gen1_path_init(struct rsnd_priv *priv, * Then, SSI id = SCU id here */ - if (rsnd_dai_is_play(rdai, io)) - id = info->ssi_id_playback; - else - id = info->ssi_id_capture; + /* get SSI's ID */ + mod = rsnd_ssi_mod_get_frm_dai(priv, + rsnd_dai_id(priv, rdai), + rsnd_dai_is_play(rdai, io)); + id = rsnd_mod_id(mod); /* SSI */ mod = rsnd_ssi_mod_get(priv, id); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0e7727cc41db..9243e387104c 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -157,6 +157,7 @@ int rsnd_dai_disconnect(struct rsnd_mod *mod); int rsnd_dai_connect(struct rsnd_dai *rdai, struct rsnd_mod *mod, struct rsnd_dai_stream *io); int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io); +int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai); #define rsnd_dai_get_platform_info(rdai) ((rdai)->info) #define rsnd_io_to_runtime(io) ((io)->substream->runtime) @@ -254,5 +255,7 @@ int rsnd_ssi_probe(struct platform_device *pdev, void rsnd_ssi_remove(struct platform_device *pdev, struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); +struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv, + int dai_id, int is_play); #endif diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 061ac7e88309..c48a6c7cd08e 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -87,6 +87,7 @@ struct rsnd_ssiu { #define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent) #define rsnd_rdai_is_clk_master(rdai) ((rdai)->clk_master) #define rsnd_ssi_mode_flags(p) ((p)->info->flags) +#define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id) #define rsnd_ssi_to_ssiu(ssi)\ (((struct rsnd_ssiu *)((ssi) - rsnd_mod_id(&(ssi)->mod))) - 1) @@ -502,6 +503,27 @@ static struct rsnd_mod_ops rsnd_ssi_non_ops = { /* * ssi mod function */ +struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv, + int dai_id, int is_play) +{ + struct rsnd_ssi *ssi; + int i, has_play; + + is_play = !!is_play; + + for_each_rsnd_ssi(ssi, priv, i) { + if (rsnd_ssi_dai_id(ssi) != dai_id) + continue; + + has_play = !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY); + + if (is_play == has_play) + return &ssi->mod; + } + + return NULL; +} + struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id) { BUG_ON(id < 0 || id >= rsnd_ssi_nr(priv)); -- cgit v1.2.3-70-g09d2 From 0a4d94c07ce782e645a8c0484d52221758b4c398 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 28 Jul 2013 18:58:50 -0700 Subject: ASoC: rsnd: add common DMAEngine method R-Car Sound driver will support DMA transfer in the future, then, SSI/SRU/SRC will use it. Current R-Car can't use soc-dmaengine-pcm.c since its DMAEngine doesn't support dmaengine_prep_dma_cyclic(), and SSI needs double plane transfer (which needs special submit) on DMAC. This patch adds common DMAEngine method for it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 132 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 32 ++++++++++++ 2 files changed, 164 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 420d6df9c3d0..a35706028514 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -173,6 +173,138 @@ void rsnd_mod_init(struct rsnd_priv *priv, INIT_LIST_HEAD(&mod->list); } +/* + * rsnd_dma functions + */ +static void rsnd_dma_continue(struct rsnd_dma *dma) +{ + /* push next A or B plane */ + dma->submit_loop = 1; + schedule_work(&dma->work); +} + +void rsnd_dma_start(struct rsnd_dma *dma) +{ + /* push both A and B plane*/ + dma->submit_loop = 2; + schedule_work(&dma->work); +} + +void rsnd_dma_stop(struct rsnd_dma *dma) +{ + dma->submit_loop = 0; + cancel_work_sync(&dma->work); + dmaengine_terminate_all(dma->chan); +} + +static void rsnd_dma_complete(void *data) +{ + struct rsnd_dma *dma = (struct rsnd_dma *)data; + struct rsnd_priv *priv = dma->priv; + unsigned long flags; + + rsnd_lock(priv, flags); + + dma->complete(dma); + + if (dma->submit_loop) + rsnd_dma_continue(dma); + + rsnd_unlock(priv, flags); +} + +static void rsnd_dma_do_work(struct work_struct *work) +{ + struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work); + struct rsnd_priv *priv = dma->priv; + struct device *dev = rsnd_priv_to_dev(priv); + struct dma_async_tx_descriptor *desc; + dma_addr_t buf; + size_t len; + int i; + + for (i = 0; i < dma->submit_loop; i++) { + + if (dma->inquiry(dma, &buf, &len) < 0) + return; + + desc = dmaengine_prep_slave_single( + dma->chan, buf, len, dma->dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); + return; + } + + desc->callback = rsnd_dma_complete; + desc->callback_param = dma; + + if (dmaengine_submit(desc) < 0) { + dev_err(dev, "dmaengine_submit() fail\n"); + return; + } + + } + + dma_async_issue_pending(dma->chan); +} + +int rsnd_dma_available(struct rsnd_dma *dma) +{ + return !!dma->chan; +} + +static bool rsnd_dma_filter(struct dma_chan *chan, void *param) +{ + chan->private = param; + + return true; +} + +int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, + int is_play, int id, + int (*inquiry)(struct rsnd_dma *dma, + dma_addr_t *buf, int *len), + int (*complete)(struct rsnd_dma *dma)) +{ + struct device *dev = rsnd_priv_to_dev(priv); + dma_cap_mask_t mask; + + if (dma->chan) { + dev_err(dev, "it already has dma channel\n"); + return -EIO; + } + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + dma->slave.shdma_slave.slave_id = id; + + dma->chan = dma_request_channel(mask, rsnd_dma_filter, + &dma->slave.shdma_slave); + if (!dma->chan) { + dev_err(dev, "can't get dma channel\n"); + return -EIO; + } + + dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; + dma->priv = priv; + dma->inquiry = inquiry; + dma->complete = complete; + INIT_WORK(&dma->work, rsnd_dma_do_work); + + return 0; +} + +void rsnd_dma_quit(struct rsnd_priv *priv, + struct rsnd_dma *dma) +{ + if (dma->chan) + dma_release_channel(dma->chan); + + dma->chan = NULL; +} + /* * rsnd_dai functions */ diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 9243e387104c..15dccd598960 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -13,9 +13,12 @@ #include #include +#include #include #include #include +#include +#include #include #include #include @@ -78,6 +81,32 @@ void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data); +/* + * R-Car DMA + */ +struct rsnd_dma { + struct rsnd_priv *priv; + struct sh_dmae_slave slave; + struct work_struct work; + struct dma_chan *chan; + enum dma_data_direction dir; + int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len); + int (*complete)(struct rsnd_dma *dma); + + int submit_loop; +}; + +void rsnd_dma_start(struct rsnd_dma *dma); +void rsnd_dma_stop(struct rsnd_dma *dma); +int rsnd_dma_available(struct rsnd_dma *dma); +int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, + int is_play, int id, + int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len), + int (*complete)(struct rsnd_dma *dma)); +void rsnd_dma_quit(struct rsnd_priv *priv, + struct rsnd_dma *dma); + + /* * R-Car sound mod */ @@ -103,9 +132,12 @@ struct rsnd_mod { struct rsnd_priv *priv; struct rsnd_mod_ops *ops; struct list_head list; /* connect to rsnd_dai playback/capture */ + struct rsnd_dma dma; }; #define rsnd_mod_to_priv(mod) ((mod)->priv) +#define rsnd_mod_to_dma(mod) (&(mod)->dma) +#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma) #define rsnd_mod_id(mod) ((mod)->id) #define for_each_rsnd_mod(pos, n, io) \ list_for_each_entry_safe(pos, n, &(io)->head, list) -- cgit v1.2.3-70-g09d2 From 849fc82a6f4f32b4c8c502bb7c4a68df51170232 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 28 Jul 2013 18:59:02 -0700 Subject: ASoC: rsnd: SSI supports DMA transfer This patch adds DMAEngine transfer on SSI. But, it transfers sound data from memory to SSI directly without using HPBIF at this time. It will be updated soon Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 7 +-- sound/soc/sh/rcar/ssi.c | 110 +++++++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 111 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index 33233edd1664..a72687dda0cd 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -39,13 +39,14 @@ #define RSND_SSI_PLAY (1 << 24) -#define RSND_SSI_SET(_dai_id, _pio_irq, _flags) \ -{ .dai_id = _dai_id, .pio_irq = _pio_irq, .flags = _flags } +#define RSND_SSI_SET(_dai_id, _dma_id, _pio_irq, _flags) \ +{ .dai_id = _dai_id, .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags } #define RSND_SSI_UNUSED \ -{ .dai_id = -1, .pio_irq = -1, .flags = 0 } +{ .dai_id = -1, .dma_id = -1, .pio_irq = -1, .flags = 0 } struct rsnd_ssi_platform_info { int dai_id; + int dma_id; int pio_irq; u32 flags; }; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index c48a6c7cd08e..2079ccf5f322 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -19,6 +19,7 @@ * SSICR */ #define FORCE (1 << 31) /* Fixed */ +#define DMEN (1 << 28) /* DMA Enable */ #define UIEN (1 << 27) /* Underflow Interrupt Enable */ #define OIEN (1 << 26) /* Overflow Interrupt Enable */ #define IIEN (1 << 25) /* Idle Mode Interrupt Enable */ @@ -51,6 +52,11 @@ #define IIRQ (1 << 25) /* Idle Mode Interrupt Status */ #define DIRQ (1 << 24) /* Data Interrupt Status Flag */ +/* + * SSIWSR + */ +#define CONT (1 << 8) /* WS Continue Function */ + struct rsnd_ssi { struct clk *clk; struct rsnd_ssi_platform_info *info; /* rcar_snd.h */ @@ -63,6 +69,7 @@ struct rsnd_ssi { u32 cr_clk; u32 cr_etc; int err; + int dma_offset; unsigned int usrcnt; unsigned int rate; }; @@ -83,7 +90,10 @@ struct rsnd_ssiu { #define rsnd_ssi_nr(priv) (((struct rsnd_ssiu *)((priv)->ssiu))->ssi_nr) #define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod) -#define rsnd_ssi_is_pio(ssi) ((ssi)->info->pio_irq > 0) +#define rsnd_dma_to_ssi(dma) rsnd_mod_to_ssi(rsnd_dma_to_mod(dma)) +#define rsnd_ssi_pio_available(ssi) ((ssi)->info->pio_irq > 0) +#define rsnd_ssi_dma_available(ssi) \ + rsnd_dma_available(rsnd_mod_to_dma(&(ssi)->mod)) #define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent) #define rsnd_rdai_is_clk_master(rdai) ((rdai)->clk_master) #define rsnd_ssi_mode_flags(p) ((p)->info->flags) @@ -477,6 +487,79 @@ static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .stop = rsnd_ssi_pio_stop, }; +static int rsnd_ssi_dma_inquiry(struct rsnd_dma *dma, dma_addr_t *buf, int *len) +{ + struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma); + struct rsnd_dai_stream *io = ssi->io; + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + + *len = io->byte_per_period; + *buf = runtime->dma_addr + + rsnd_dai_pointer_offset(io, ssi->dma_offset + *len); + ssi->dma_offset = *len; /* it cares A/B plane */ + + return 0; +} + +static int rsnd_ssi_dma_complete(struct rsnd_dma *dma) +{ + struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma); + struct rsnd_dai_stream *io = ssi->io; + u32 status = rsnd_mod_read(&ssi->mod, SSISR); + + rsnd_ssi_record_error(ssi, status); + + rsnd_dai_pointer_update(ssi->io, io->byte_per_period); + + return 0; +} + +static int rsnd_ssi_dma_start(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod); + + /* enable DMA transfer */ + ssi->cr_etc = DMEN; + ssi->dma_offset = 0; + + rsnd_dma_start(dma); + + rsnd_ssi_hw_start(ssi, ssi->rdai, io); + + /* enable WS continue */ + if (rsnd_rdai_is_clk_master(rdai)) + rsnd_mod_write(&ssi->mod, SSIWSR, CONT); + + return 0; +} + +static int rsnd_ssi_dma_stop(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod); + + ssi->cr_etc = 0; + + rsnd_ssi_hw_stop(ssi, rdai); + + rsnd_dma_stop(dma); + + return 0; +} + +static struct rsnd_mod_ops rsnd_ssi_dma_ops = { + .name = "ssi (dma)", + .init = rsnd_ssi_init, + .quit = rsnd_ssi_quit, + .start = rsnd_ssi_dma_start, + .stop = rsnd_ssi_dma_stop, +}; + /* * Non SSI */ @@ -573,10 +656,27 @@ int rsnd_ssi_probe(struct platform_device *pdev, ops = &rsnd_ssi_non_ops; + /* + * SSI DMA case + */ + if (pinfo->dma_id > 0) { + ret = rsnd_dma_init( + priv, rsnd_mod_to_dma(&ssi->mod), + (rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY), + pinfo->dma_id, + rsnd_ssi_dma_inquiry, + rsnd_ssi_dma_complete); + if (ret < 0) + dev_info(dev, "SSI DMA failed. try PIO transter\n"); + else + ops = &rsnd_ssi_dma_ops; + } + /* * SSI PIO case */ - if (rsnd_ssi_is_pio(ssi)) { + if (!rsnd_ssi_dma_available(ssi) && + rsnd_ssi_pio_available(ssi)) { ret = devm_request_irq(dev, pinfo->pio_irq, &rsnd_ssi_pio_interrupt, IRQF_SHARED, @@ -605,6 +705,10 @@ void rsnd_ssi_remove(struct platform_device *pdev, struct rsnd_ssi *ssi; int i; - for_each_rsnd_ssi(ssi, priv, i) + for_each_rsnd_ssi(ssi, priv, i) { clk_put(ssi->clk); + if (rsnd_ssi_dma_available(ssi)) + rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod)); + } + } -- cgit v1.2.3-70-g09d2 From 374a528111fa07878090bd9694a3e153814de39c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 28 Jul 2013 18:59:12 -0700 Subject: ASoC: rsnd: SSI supports DMA transfer via BUSIF This patch adds BUSIF support for R-Car sound DMAEngine transfer. The sound data will be transferred via FIFO which can cover blank time which will happen when DMA channel is switching. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 6 ++ sound/soc/sh/rcar/gen.c | 10 ++- sound/soc/sh/rcar/rsnd.h | 9 +++ sound/soc/sh/rcar/scu.c | 154 ++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/sh/rcar/ssi.c | 18 +++++- 5 files changed, 190 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index a72687dda0cd..d35412ae03b3 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -36,6 +36,7 @@ #define RSND_SSI_CLK_PIN_SHARE (1 << 31) #define RSND_SSI_CLK_FROM_ADG (1 << 30) /* clock parent is master */ #define RSND_SSI_SYNC (1 << 29) /* SSI34_sync etc */ +#define RSND_SSI_DEPENDENT (1 << 28) /* SSI needs SRU/SCU */ #define RSND_SSI_PLAY (1 << 24) @@ -51,6 +52,11 @@ struct rsnd_ssi_platform_info { u32 flags; }; +/* + * flags + */ +#define RSND_SCU_USB_HPBIF (1 << 31) /* it needs RSND_SSI_DEPENDENT */ + struct rsnd_scu_platform_info { u32 flags; }; diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 460c57eef267..babb203b43b7 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -34,9 +34,6 @@ struct rsnd_gen { #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) -#define rsnd_is_gen1(s) ((s)->info->flags & RSND_GEN1) -#define rsnd_is_gen2(s) ((s)->info->flags & RSND_GEN2) - /* * Gen2 * will be filled in the future @@ -115,8 +112,15 @@ static struct rsnd_gen_ops rsnd_gen1_ops = { static void rsnd_gen1_reg_map_init(struct rsnd_gen *gen) { + RSND_GEN1_REG_MAP(gen, SRU, SRC_ROUTE_SEL, 0x0, 0x00); + RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL0, 0x0, 0x08); + RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL1, 0x0, 0x0c); + RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL2, 0x0, 0x10); + RSND_GEN1_REG_MAP(gen, SRU, SRC_CTRL, 0x0, 0xc0); RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE0, 0x0, 0xD0); RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE1, 0x0, 0xD4); + RSND_GEN1_REG_MAP(gen, SRU, BUSIF_MODE, 0x4, 0x20); + RSND_GEN1_REG_MAP(gen, SRU, BUSIF_ADINR, 0x40, 0x214); RSND_GEN1_REG_MAP(gen, ADG, BRRA, 0x0, 0x00); RSND_GEN1_REG_MAP(gen, ADG, BRRB, 0x0, 0x04); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 15dccd598960..9cc6986a8cfb 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -32,8 +32,15 @@ */ enum rsnd_reg { /* SRU/SCU */ + RSND_REG_SRC_ROUTE_SEL, + RSND_REG_SRC_TMG_SEL0, + RSND_REG_SRC_TMG_SEL1, + RSND_REG_SRC_TMG_SEL2, + RSND_REG_SRC_CTRL, RSND_REG_SSI_MODE0, RSND_REG_SSI_MODE1, + RSND_REG_BUSIF_MODE, + RSND_REG_BUSIF_ADINR, /* ADG */ RSND_REG_BRRA, @@ -213,6 +220,8 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); +#define rsnd_is_gen1(s) ((s)->info->flags & RSND_GEN1) +#define rsnd_is_gen2(s) ((s)->info->flags & RSND_GEN2) /* * R-Car ADG diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index c12e65f240a1..29837e326bc5 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -15,6 +15,18 @@ struct rsnd_scu { struct rsnd_mod mod; }; +#define rsnd_scu_mode_flags(p) ((p)->info->flags) + +/* + * ADINR + */ +#define OTBL_24 (0 << 16) +#define OTBL_22 (2 << 16) +#define OTBL_20 (4 << 16) +#define OTBL_18 (6 << 16) +#define OTBL_16 (8 << 16) + + #define rsnd_mod_to_scu(_mod) \ container_of((_mod), struct rsnd_scu, mod) @@ -24,6 +36,116 @@ struct rsnd_scu { ((pos) = (struct rsnd_scu *)(priv)->scu + i); \ i++) +static int rsnd_scu_set_route(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct scu_route_config { + u32 mask; + int shift; + } routes[] = { + { 0xF, 0, }, /* 0 */ + { 0xF, 4, }, /* 1 */ + { 0xF, 8, }, /* 2 */ + { 0x7, 12, }, /* 3 */ + { 0x7, 16, }, /* 4 */ + { 0x7, 20, }, /* 5 */ + { 0x7, 24, }, /* 6 */ + { 0x3, 28, }, /* 7 */ + { 0x3, 30, }, /* 8 */ + }; + + u32 mask; + u32 val; + int shift; + int id; + + /* + * Gen1 only + */ + if (!rsnd_is_gen1(priv)) + return 0; + + id = rsnd_mod_id(mod); + if (id < 0 || id > ARRAY_SIZE(routes)) + return -EIO; + + /* + * SRC_ROUTE_SELECT + */ + val = rsnd_dai_is_play(rdai, io) ? 0x1 : 0x2; + val = val << routes[id].shift; + mask = routes[id].mask << routes[id].shift; + + rsnd_mod_bset(mod, SRC_ROUTE_SEL, mask, val); + + /* + * SRC_TIMING_SELECT + */ + shift = (id % 4) * 8; + mask = 0x1F << shift; + if (8 == id) /* SRU8 is very special */ + val = id << shift; + else + val = (id + 1) << shift; + + switch (id / 4) { + case 0: + rsnd_mod_bset(mod, SRC_TMG_SEL0, mask, val); + break; + case 1: + rsnd_mod_bset(mod, SRC_TMG_SEL1, mask, val); + break; + case 2: + rsnd_mod_bset(mod, SRC_TMG_SEL2, mask, val); + break; + } + + return 0; +} + +static int rsnd_scu_set_mode(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + int id = rsnd_mod_id(mod); + u32 val; + + if (rsnd_is_gen1(priv)) { + val = (1 << id); + rsnd_mod_bset(mod, SRC_CTRL, val, val); + } + + return 0; +} + +static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + u32 adinr = runtime->channels; + + switch (runtime->sample_bits) { + case 16: + adinr |= OTBL_16; + break; + case 32: + adinr |= OTBL_24; + break; + default: + return -EIO; + } + + rsnd_mod_write(mod, BUSIF_MODE, 1); + rsnd_mod_write(mod, BUSIF_ADINR, adinr); + + return 0; +} + static int rsnd_scu_init(struct rsnd_mod *mod, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) @@ -53,9 +175,36 @@ static int rsnd_scu_start(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); struct device *dev = rsnd_priv_to_dev(priv); + u32 flags = rsnd_scu_mode_flags(scu); + int ret; + + /* + * SCU will be used if it has RSND_SCU_USB_HPBIF flags + */ + if (!(flags & RSND_SCU_USB_HPBIF)) { + /* it use PIO transter */ + dev_dbg(dev, "%s%d is not used\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return 0; + } + + /* it use DMA transter */ + ret = rsnd_scu_set_route(priv, mod, rdai, io); + if (ret < 0) + return ret; + + ret = rsnd_scu_set_mode(priv, mod, rdai, io); + if (ret < 0) + return ret; - dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + ret = rsnd_scu_set_hpbif(priv, mod, rdai, io); + if (ret < 0) + return ret; + + dev_dbg(dev, "%s%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); return 0; } @@ -112,8 +261,9 @@ int rsnd_scu_probe(struct platform_device *pdev, rsnd_mod_init(priv, &scu->mod, &rsnd_scu_ops, i); scu->info = &info->scu_info[i]; - } + dev_dbg(dev, "SCU%d probed\n", i); + } dev_dbg(dev, "scu probed\n"); return 0; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 2079ccf5f322..fae26d3f79d2 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -104,6 +104,7 @@ struct rsnd_ssiu { static void rsnd_ssi_mode_init(struct rsnd_priv *priv, struct rsnd_ssiu *ssiu) { + struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_ssi *ssi; u32 flags; u32 val; @@ -113,8 +114,17 @@ static void rsnd_ssi_mode_init(struct rsnd_priv *priv, * SSI_MODE0 */ ssiu->ssi_mode0 = 0; - for_each_rsnd_ssi(ssi, priv, i) - ssiu->ssi_mode0 |= (1 << i); + for_each_rsnd_ssi(ssi, priv, i) { + flags = rsnd_ssi_mode_flags(ssi); + + /* see also BUSIF_MODE */ + if (!(flags & RSND_SSI_DEPENDENT)) { + ssiu->ssi_mode0 |= (1 << i); + dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", i); + } else { + dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", i); + } + } /* * SSI_MODE1 @@ -670,6 +680,8 @@ int rsnd_ssi_probe(struct platform_device *pdev, dev_info(dev, "SSI DMA failed. try PIO transter\n"); else ops = &rsnd_ssi_dma_ops; + + dev_dbg(dev, "SSI%d use DMA transfer\n", i); } /* @@ -687,6 +699,8 @@ int rsnd_ssi_probe(struct platform_device *pdev, } ops = &rsnd_ssi_pio_ops; + + dev_dbg(dev, "SSI%d use PIO transfer\n", i); } rsnd_mod_init(priv, &ssi->mod, ops, i); -- cgit v1.2.3-70-g09d2 From 2460719c79854a3bebe569cbfbfa0b1caa1dc434 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 28 Jul 2013 18:59:25 -0700 Subject: ASoC: rsnd: scu: cleanup empty functions This patch cleanups empty functions on scu Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/scu.c | 39 --------------------------------------- 1 file changed, 39 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 29837e326bc5..184d9008cecd 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -146,30 +146,6 @@ static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, return 0; } -static int rsnd_scu_init(struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) -{ - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct device *dev = rsnd_priv_to_dev(priv); - - dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); - - return 0; -} - -static int rsnd_scu_quit(struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) -{ - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct device *dev = rsnd_priv_to_dev(priv); - - dev_dbg(dev, "%s.%d quit\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); - - return 0; -} - static int rsnd_scu_start(struct rsnd_mod *mod, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) @@ -209,24 +185,9 @@ static int rsnd_scu_start(struct rsnd_mod *mod, return 0; } -static int rsnd_scu_stop(struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) -{ - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct device *dev = rsnd_priv_to_dev(priv); - - dev_dbg(dev, "%s.%d stop\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); - - return 0; -} - static struct rsnd_mod_ops rsnd_scu_ops = { .name = "scu", - .init = rsnd_scu_init, - .quit = rsnd_scu_quit, .start = rsnd_scu_start, - .stop = rsnd_scu_stop, }; struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id) -- cgit v1.2.3-70-g09d2 From 8548a464b942a97324d0e3e340ce95356cff32c4 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Sat, 27 Jul 2013 13:31:52 +0200 Subject: ASoC: imx-audmux: Read default configuration from devicetree Adds a function to parse a default port configuration from devicetree. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/imx-audmux.txt | 9 ++++ sound/soc/fsl/imx-audmux.c | 62 ++++++++++++++++++++++ 2 files changed, 71 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt index 215aa9817213..f88a00e54c63 100644 --- a/Documentation/devicetree/bindings/sound/imx-audmux.txt +++ b/Documentation/devicetree/bindings/sound/imx-audmux.txt @@ -5,6 +5,15 @@ Required properties: or "fsl,imx31-audmux" for the version firstly used on i.MX31. - reg : Should contain AUDMUX registers location and length +An initial configuration can be setup using child nodes. + +Required properties of optional child nodes: +- fsl,audmux-port : Integer of the audmux port that is configured by this + child node. +- fsl,port-config : List of configuration options for the specific port. For + imx31-audmux and above, it is a list of tuples . For + imx21-audmux it is a list of pcr values. + Example: audmux@021d8000 { diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 1a5da1e13077..103d1b020496 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -251,6 +251,66 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, } EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); +static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, + struct device_node *of_node) +{ + struct device_node *child; + + for_each_available_child_of_node(of_node, child) { + unsigned int port; + unsigned int ptcr = 0; + unsigned int pdcr = 0; + unsigned int pcr = 0; + unsigned int val; + int ret; + int i = 0; + + ret = of_property_read_u32(child, "fsl,audmux-port", &port); + if (ret) { + dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n", + child->full_name); + continue; + } + if (!of_property_read_bool(child, "fsl,port-config")) { + dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n", + child->full_name); + continue; + } + + for (i = 0; (ret = of_property_read_u32_index(child, + "fsl,port-config\n", i, &val)) == 0; + ++i) { + if (audmux_type == IMX31_AUDMUX) { + if (i % 2) + pdcr |= val; + else + ptcr |= val; + } else { + pcr |= val; + } + } + + if (ret != -ENODATA) { + dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n", + i, child->full_name); + continue; + } + + if (audmux_type == IMX31_AUDMUX) { + if (i % 2) { + dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n", + child->full_name); + continue; + } + imx_audmux_v2_configure_port(port, ptcr, pdcr); + } else { + imx_audmux_v1_configure_port(port, pcr); + } + } + + return 0; +} + static int imx_audmux_probe(struct platform_device *pdev) { struct resource *res; @@ -275,6 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev) if (audmux_type == IMX31_AUDMUX) audmux_debugfs_init(); + imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); + return 0; } -- cgit v1.2.3-70-g09d2 From de623ece5be03e4447dbe08eaca30c92202a34a2 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Sat, 27 Jul 2013 13:31:53 +0200 Subject: ASoC: fsl-ssi: Add support for imx-pcm-fiq Add support for non-dma pcm for imx platforms with imx-pcm-fiq support. Instead of imx-pcm-audio, in this case imx-pcm-fiq-audio device is added and the SIER flags are set differently. We need imx-pcm-fiq for some boards that use an incompatible codec. imx-pcm-fiq handles those codecs differently and allows to operate with them. DMA is not possible because some data sent by the codecs, e.g. wm9712, is not in the datastream. Also some data is mixed up in the fifos, so that we need to sort them out manually. Signed-off-by: Markus Pargmann Tested-by: Shawn Guo Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,ssi.txt | 4 ++ sound/soc/fsl/fsl_ssi.c | 79 ++++++++++++++++++---- 2 files changed, 71 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt index 5ff76c9c57d2..e45cbce9cbf3 100644 --- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt @@ -47,6 +47,10 @@ Optional properties: - codec-handle: Phandle to a 'codec' node that defines an audio codec connected to this SSI. This node is typically a child of an I2C or other control node. +- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to + filter the codec stream. This is necessary for some boards + where an incompatible codec is connected to this SSI, e.g. + on pca100 and pcm043. Child 'codec' node required properties: - compatible: Compatible list, contains the name of the codec diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4d78df7d7f34..8b075ef5c6b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -8,6 +8,26 @@ * This file is licensed under the terms of the GNU General Public License * version 2. This program is licensed "as is" without any warranty of any * kind, whether express or implied. + * + * + * Some notes why imx-pcm-fiq is used instead of DMA on some boards: + * + * The i.MX SSI core has some nasty limitations in AC97 mode. While most + * sane processor vendors have a FIFO per AC97 slot, the i.MX has only + * one FIFO which combines all valid receive slots. We cannot even select + * which slots we want to receive. The WM9712 with which this driver + * was developed with always sends GPIO status data in slot 12 which + * we receive in our (PCM-) data stream. The only chance we have is to + * manually skip this data in the FIQ handler. With sampling rates different + * from 48000Hz not every frame has valid receive data, so the ratio + * between pcm data and GPIO status data changes. Our FIQ handler is not + * able to handle this, hence this driver only works with 48000Hz sampling + * rate. + * Reading and writing AC97 registers is another challenge. The core + * provides us status bits when the read register is updated with *another* + * value. When we read the same register two times (and the register still + * contains the same value) these status bits are not set. We work + * around this by not polling these bits but only wait a fixed delay. */ #include @@ -121,11 +141,13 @@ struct fsl_ssi_private { bool new_binding; bool ssi_on_imx; + bool use_dma; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; struct imx_dma_data filter_data_tx; struct imx_dma_data filter_data_rx; + struct imx_pcm_fiq_params fiq_params; struct { unsigned int rfrc; @@ -355,7 +377,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ /* Enable the interrupts and DMA requests */ - write_ssi(SIER_FLAGS, &ssi->sier); + if (ssi_private->use_dma) + write_ssi(SIER_FLAGS, &ssi->sier); + else + write_ssi(CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN | + CCSR_SSI_SIER_RIE | + CCSR_SSI_SIER_RFF0_EN, &ssi->sier); /* * Set the watermark for transmit FIFI 0 and receive FIFO 0. We @@ -543,7 +570,7 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai); - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx && ssi_private->use_dma) { dai->playback_dma_data = &ssi_private->dma_params_tx; dai->capture_dma_data = &ssi_private->dma_params_rx; } @@ -683,6 +710,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) strcpy(ssi_private->name, p); + ssi_private->use_dma = !of_property_read_bool(np, + "fsl,fiq-stream-filter"); + /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, sizeof(fsl_ssi_dai_template)); @@ -707,12 +737,16 @@ static int fsl_ssi_probe(struct platform_device *pdev) return -ENXIO; } - /* The 'name' should not have any slashes in it. */ - ret = devm_request_irq(&pdev->dev, ssi_private->irq, fsl_ssi_isr, 0, - ssi_private->name, ssi_private); - if (ret < 0) { - dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); - goto error_irqmap; + if (ssi_private->use_dma) { + /* The 'name' should not have any slashes in it. */ + ret = devm_request_irq(&pdev->dev, ssi_private->irq, + fsl_ssi_isr, 0, ssi_private->name, + ssi_private); + if (ret < 0) { + dev_err(&pdev->dev, "could not claim irq %u\n", + ssi_private->irq); + goto error_irqmap; + } } /* Are the RX and the TX clocks locked? */ @@ -766,7 +800,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ret = of_property_read_u32_array(pdev->dev.of_node, "fsl,ssi-dma-events", dma_events, 2); - if (ret) { + if (ret && !ssi_private->use_dma) { dev_err(&pdev->dev, "could not get dma events\n"); goto error_clk; } @@ -805,9 +839,30 @@ static int fsl_ssi_probe(struct platform_device *pdev) } if (ssi_private->ssi_on_imx) { - ret = imx_pcm_dma_init(pdev); - if (ret) - goto error_dev; + if (!ssi_private->use_dma) { + + /* + * Some boards use an incompatible codec. To get it + * working, we are using imx-fiq-pcm-audio, that + * can handle those codecs. DMA is not possible in this + * situation. + */ + + ssi_private->fiq_params.irq = ssi_private->irq; + ssi_private->fiq_params.base = ssi_private->ssi; + ssi_private->fiq_params.dma_params_rx = + &ssi_private->dma_params_rx; + ssi_private->fiq_params.dma_params_tx = + &ssi_private->dma_params_tx; + + ret = imx_pcm_fiq_init(pdev, &ssi_private->fiq_params); + if (ret) + goto error_dev; + } else { + ret = imx_pcm_dma_init(pdev); + if (ret) + goto error_dev; + } } /* -- cgit v1.2.3-70-g09d2 From 3a5e517bb2e9856fd836e90caa415f116d34bd04 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Sat, 27 Jul 2013 13:31:54 +0200 Subject: ASoC: fsl-ssi: Use generic DMA bindings if possible There may be some platforms using fsl-ssi that do not have a DMA driver with generic DMA bindings. So this patch adds support for the generic DMA bindings, while still accepting the old "fsl,dma-events" property if "dmas" is not found. Signed-off-by: Markus Pargmann Tested-by: Shawn Guo Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl,ssi.txt | 4 ++++ sound/soc/fsl/fsl_ssi.c | 20 ++++++++++++-------- 2 files changed, 16 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt index e45cbce9cbf3..088a2c038f01 100644 --- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt @@ -51,6 +51,10 @@ Optional properties: filter the codec stream. This is necessary for some boards where an incompatible codec is connected to this SSI, e.g. on pca100 and pcm043. +- dmas: Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. +- dma-names: Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq + is not defined. Child 'codec' node required properties: - compatible: Compatible list, contains the name of the codec diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8b075ef5c6b9..0c072ff10875 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -794,15 +794,19 @@ static int fsl_ssi_probe(struct platform_device *pdev) &ssi_private->filter_data_tx; ssi_private->dma_params_rx.filter_data = &ssi_private->filter_data_rx; - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32_array(pdev->dev.of_node, + if (!of_property_read_bool(pdev->dev.of_node, "dmas") && + ssi_private->use_dma) { + /* + * FIXME: This is a temporary solution until all + * necessary dma drivers support the generic dma + * bindings. + */ + ret = of_property_read_u32_array(pdev->dev.of_node, "fsl,ssi-dma-events", dma_events, 2); - if (ret && !ssi_private->use_dma) { - dev_err(&pdev->dev, "could not get dma events\n"); - goto error_clk; + if (ret && ssi_private->use_dma) { + dev_err(&pdev->dev, "could not get dma events but fsl-ssi is configured to use DMA\n"); + goto error_clk; + } } shared = of_device_is_compatible(of_get_parent(np), -- cgit v1.2.3-70-g09d2 From fdbcb3cba54b29a37dfe42acdc0e72c543e0807d Mon Sep 17 00:00:00 2001 From: Nicolas Ferre Date: Tue, 30 Jul 2013 12:32:03 +0200 Subject: ASoC: atmel: machine driver for at91sam9x5-wm8731 boards Description of the Asoc machine driver for an at91sam9x5 based board with a wm8731 audio DAC. Wm8731 is clocked by a crystal and used as a master on the SSC/I2S interface. Its connections are a headphone jack and an Line input jack. [Richard: this is based on an old patch from Nicolas that I forward ported and reworked to use only device tree] Signed-off-by: Nicolas Ferre Signed-off-by: Richard Genoud Signed-off-by: Mark Brown --- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 35 ++++ Documentation/devicetree/bindings/sound/wm8731.txt | 9 + sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 208 +++++++++++++++++++++ 5 files changed, 264 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 sound/soc/atmel/sam9x5_wm8731.c (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt new file mode 100644 index 000000000000..0720857089a7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt @@ -0,0 +1,35 @@ +* Atmel at91sam9x5ek wm8731 audio complex + +Required properties: + - compatible: "atmel,sam9x5-wm8731-audio" + - atmel,model: The user-visible name of this sound complex. + - atmel,ssc-controller: The phandle of the SSC controller + - atmel,audio-codec: The phandle of the WM8731 audio codec + - atmel,audio-routing: A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headphone Jack + * Line In Jack + +wm8731 pins: +cf Documentation/devicetree/bindings/sound/wm8731.txt + +Example: +sound { + compatible = "atmel,sam9x5-wm8731-audio"; + + atmel,model = "wm8731 @ AT91SAM9X5EK"; + + atmel,audio-routing = + "Headphone Jack", "RHPOUT", + "Headphone Jack", "LHPOUT", + "LLINEIN", "Line In Jack", + "RLINEIN", "Line In Jack"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8731>; +}; diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt index 15f70048469b..236690e99b87 100644 --- a/Documentation/devicetree/bindings/sound/wm8731.txt +++ b/Documentation/devicetree/bindings/sound/wm8731.txt @@ -16,3 +16,12 @@ codec: wm8731@1a { compatible = "wlf,wm8731"; reg = <0x1a>; }; + +Available audio endpoints for an audio-routing table: + * LOUT: Left Channel Line Output + * ROUT: Right Channel Line Output + * LHPOUT: Left Channel Headphone Output + * RHPOUT: Right Channel Headphone Output + * LLINEIN: Left Channel Line Input + * RLINEIN: Right Channel Line Input + * MICIN: Microphone Input diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 986323b4caad..e48d38a1b95c 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -43,6 +43,16 @@ config SND_ATMEL_SOC_WM8904 Say Y if you want to add support for Atmel ASoC driver for boards using WM8904 codec. +config SND_AT91_SOC_SAM9X5_WM8731 + tristate "SoC Audio support for WM8731-based at91sam9x5 board" + depends on ATMEL_SSC && SND_ATMEL_SOC && SOC_AT91SAM9X5 + select SND_ATMEL_SOC_SSC + select SND_ATMEL_SOC_DMA + select SND_SOC_WM8731 + help + Say Y if you want to add support for audio SoC on an + at91sam9x5 based board that is using WM8731 codec. + config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 922d4da57109..5baabc8bde3a 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -12,7 +12,9 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o snd-atmel-soc-wm8904-objs := atmel_wm8904.o +snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o +obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c new file mode 100644 index 000000000000..992ae38d5a15 --- /dev/null +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -0,0 +1,208 @@ +/* + * sam9x5_wm8731 -- SoC audio for AT91SAM9X5-based boards + * that are using WM8731 as codec. + * + * Copyright (C) 2011 Atmel, + * Nicolas Ferre + * + * Copyright (C) 2013 Paratronic, + * Richard Genoud + * + * Based on sam9g20_wm8731.c by: + * Sedji Gaouaou + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "../codecs/wm8731.h" +#include "atmel_ssc_dai.h" + + +#define MCLK_RATE 12288000 + +#define DRV_NAME "sam9x5-snd-wm8731" + +struct sam9x5_drvdata { + int ssc_id; +}; + +/* + * Logic for a wm8731 as connected on a at91sam9x5ek based board. + */ +static int sam9x5_wm8731_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct device *dev = rtd->dev; + int ret; + + dev_dbg(dev, "ASoC: %s called\n", __func__); + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, + MCLK_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to set WM8731 SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +/* + * Audio paths on at91sam9x5ek board: + * + * |A| ------------> | | ---R----> Headphone Jack + * |T| <----\ | WM | ---L--/ + * |9| ---> CLK <--> | 8731 | <--R----- Line In Jack + * |1| <------------ | | <--L--/ + */ +static const struct snd_soc_dapm_widget sam9x5_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), +}; + +static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *codec_np, *cpu_np; + struct snd_soc_card *card; + struct snd_soc_dai_link *dai; + struct sam9x5_drvdata *priv; + int ret; + + if (!np) { + dev_err(&pdev->dev, "No device node supplied\n"); + return -EINVAL; + } + + card = devm_kzalloc(&pdev->dev, sizeof(*card), GFP_KERNEL); + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL); + if (!dai || !card || !priv) { + ret = -ENOMEM; + goto out; + } + + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dai_link = dai; + card->num_links = 1; + card->dapm_widgets = sam9x5_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sam9x5_dapm_widgets); + dai->name = "WM8731"; + dai->stream_name = "WM8731 PCM"; + dai->codec_dai_name = "wm8731-hifi"; + dai->init = sam9x5_wm8731_init; + dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM; + + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) { + dev_err(&pdev->dev, "atmel,model node missing\n"); + goto out; + } + + ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "atmel,audio-routing node missing\n"); + goto out; + } + + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "atmel,audio-codec node missing\n"); + ret = -EINVAL; + goto out; + } + + dai->codec_of_node = codec_np; + + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "atmel,ssc-controller node missing\n"); + ret = -EINVAL; + goto out; + } + dai->cpu_of_node = cpu_np; + dai->platform_of_node = cpu_np; + + priv->ssc_id = of_alias_get_id(cpu_np, "ssc"); + + ret = atmel_ssc_set_audio(priv->ssc_id); + if (ret != 0) { + dev_err(&pdev->dev, + "ASoC: Failed to set SSC %d for audio: %d\n", + ret, priv->ssc_id); + goto out; + } + + of_node_put(codec_np); + of_node_put(cpu_np); + + platform_set_drvdata(pdev, card); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, + "ASoC: Platform device allocation failed\n"); + goto out_put_audio; + } + + dev_dbg(&pdev->dev, "ASoC: %s ok\n", __func__); + + return ret; + +out_put_audio: + atmel_ssc_put_audio(priv->ssc_id); +out: + return ret; +} + +static int sam9x5_wm8731_driver_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct sam9x5_drvdata *priv = card->drvdata; + + snd_soc_unregister_card(card); + atmel_ssc_put_audio(priv->ssc_id); + + return 0; +} + +static const struct of_device_id sam9x5_wm8731_of_match[] = { + { .compatible = "atmel,sam9x5-wm8731-audio", }, + {}, +}; +MODULE_DEVICE_TABLE(of, sam9x5_wm8731_of_match); + +static struct platform_driver sam9x5_wm8731_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(sam9x5_wm8731_of_match), + }, + .probe = sam9x5_wm8731_driver_probe, + .remove = sam9x5_wm8731_driver_remove, +}; +module_platform_driver(sam9x5_wm8731_driver); + +/* Module information */ +MODULE_AUTHOR("Nicolas Ferre "); +MODULE_AUTHOR("Richard Genoud "); +MODULE_DESCRIPTION("ALSA SoC machine driver for AT91SAM9x5 - WM8731"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- cgit v1.2.3-70-g09d2 From e2c98a8bba958045bde861fe1d66be54315c7790 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 6 Aug 2013 12:57:21 -0500 Subject: ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume Beep Volume Min/Max was backwards. Change to SOC_SONGLE_SX_TLV for correct volume representation Signed-off-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 987f728718c5..ee25f325d65c 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -451,7 +451,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- cgit v1.2.3-70-g09d2 From 8806d96db7b04fffba4cfc9ceac31d24c8517fe9 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 6 Aug 2013 12:57:22 -0500 Subject: ASoC: cs42l52: Add new TLV for Beep Volume CS42L52 Beep control uses 2dB scale from -56dB Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index ee25f325d65c..be2ba1b6fe4a 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -195,6 +195,8 @@ static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); + static const unsigned int limiter_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), @@ -451,7 +453,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, + 0, 0x07, 0x1f, beep_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- cgit v1.2.3-70-g09d2 From 9f19de649f70c3bd32da09fc08643d4fca1d06fe Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 6 Aug 2013 18:03:07 -0300 Subject: ASoC: imx-mc13783: Make SND_SOC_IMX_MC13783 visible again Commit 02502da45 (ASoC: imx-mc13783: Depend on ARCH_ARM) introduced 'ARCH_ARM' as a dependency for SND_SOC_IMX_MC13783, but this is a non-existent symbol. This makes the selection of SND_SOC_IMX_MC13783 to be impossible. Use the correct 'ARM' symbol instead. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index c26449b54270..e15f77197d0b 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -194,7 +194,7 @@ config SND_SOC_IMX_SGTL5000 config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" - depends on MFD_MC13783 && ARCH_ARM + depends on MFD_MC13783 && ARM select SND_SOC_IMX_SSI select SND_SOC_IMX_AUDMUX select SND_SOC_MC13783 -- cgit v1.2.3-70-g09d2 From 9e7e474c0963dfd1f60b200160ff9e7cefb32b06 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Aug 2013 23:51:12 +0100 Subject: ASoC: ad1980: Provide stub DAPM support Since non-DAPM devices are not going to be supported provide DAPM input and output widgets and hook them up to the DAIs. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/ad1980.c | 43 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 43 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 89fcf7d6e7b8..7257a8885f42 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -96,6 +96,44 @@ SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; +static const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_INPUT("CD_L"), +SND_SOC_DAPM_INPUT("CD_R"), +SND_SOC_DAPM_INPUT("AUX_L"), +SND_SOC_DAPM_INPUT("AUX_R"), +SND_SOC_DAPM_INPUT("LINE_IN_L"), +SND_SOC_DAPM_INPUT("LINE_IN_R"), + +SND_SOC_DAPM_OUTPUT("LFE_OUT"), +SND_SOC_DAPM_OUTPUT("CENTER_OUT"), +SND_SOC_DAPM_OUTPUT("LINE_OUT_L"), +SND_SOC_DAPM_OUTPUT("LINE_OUT_R"), +SND_SOC_DAPM_OUTPUT("MONO_OUT"), +SND_SOC_DAPM_OUTPUT("HP_OUT_L"), +SND_SOC_DAPM_OUTPUT("HP_OUT_R"), +}; + +static const struct snd_soc_dapm_route ad1980_dapm_routes[] = { + { "Capture", NULL, "MIC1" }, + { "Capture", NULL, "MIC2" }, + { "Capture", NULL, "CD_L" }, + { "Capture", NULL, "CD_R" }, + { "Capture", NULL, "AUX_L" }, + { "Capture", NULL, "AUX_R" }, + { "Capture", NULL, "LINE_IN_L" }, + { "Capture", NULL, "LINE_IN_R" }, + + { "LFE_OUT", NULL, "Playback" }, + { "CENTER_OUT", NULL, "Playback" }, + { "LINE_OUT_L", NULL, "Playback" }, + { "LINE_OUT_R", NULL, "Playback" }, + { "MONO_OUT", NULL, "Playback" }, + { "HP_OUT_L", NULL, "Playback" }, + { "HP_OUT_R", NULL, "Playback" }, +}; + static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -253,6 +291,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1980 = { .reg_cache_step = 2, .write = ac97_write, .read = ac97_read, + + .dapm_widgets = ad1980_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets), + .dapm_routes = ad1980_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad1980_dapm_routes), }; static int ad1980_probe(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From ddb6b5a964371e8e52e696b2b258bda144c8bd3f Mon Sep 17 00:00:00 2001 From: Jussi Kivilinna Date: Tue, 6 Aug 2013 14:53:24 +0300 Subject: ALSA: 6fire: fix DMA issues with URB transfer_buffer usage Patch fixes 6fire not to use stack as URB transfer_buffer. URB buffers need to be DMA-able, which stack is not. Furthermore, transfer_buffer should not be allocated as part of larger device structure because DMA coherency issues and patch fixes this issue too. Cc: stable@vger.kernel.org Signed-off-by: Jussi Kivilinna Tested-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/comm.c | 38 +++++++++++++++++++++++++++++++++----- sound/usb/6fire/comm.h | 2 +- 2 files changed, 34 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c index 9e6e3ffd86bb..23452ee617e1 100644 --- a/sound/usb/6fire/comm.c +++ b/sound/usb/6fire/comm.c @@ -110,19 +110,37 @@ static int usb6fire_comm_send_buffer(u8 *buffer, struct usb_device *dev) static int usb6fire_comm_write8(struct comm_runtime *rt, u8 request, u8 reg, u8 value) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, value, 0x00); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } static int usb6fire_comm_write16(struct comm_runtime *rt, u8 request, u8 reg, u8 vl, u8 vh) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, vl, vh); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } int usb6fire_comm_init(struct sfire_chip *chip) @@ -135,6 +153,12 @@ int usb6fire_comm_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + rt->receiver_buffer = kzalloc(COMM_RECEIVER_BUFSIZE, GFP_KERNEL); + if (!rt->receiver_buffer) { + kfree(rt); + return -ENOMEM; + } + urb = &rt->receiver; rt->serial = 1; rt->chip = chip; @@ -153,6 +177,7 @@ int usb6fire_comm_init(struct sfire_chip *chip) urb->interval = 1; ret = usb_submit_urb(urb, GFP_KERNEL); if (ret < 0) { + kfree(rt->receiver_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "cannot create comm data receiver."); return ret; @@ -171,6 +196,9 @@ void usb6fire_comm_abort(struct sfire_chip *chip) void usb6fire_comm_destroy(struct sfire_chip *chip) { - kfree(chip->comm); + struct comm_runtime *rt = chip->comm; + + kfree(rt->receiver_buffer); + kfree(rt); chip->comm = NULL; } diff --git a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h index 6a0840b0dcff..780d5ed8e5d8 100644 --- a/sound/usb/6fire/comm.h +++ b/sound/usb/6fire/comm.h @@ -24,7 +24,7 @@ struct comm_runtime { struct sfire_chip *chip; struct urb receiver; - u8 receiver_buffer[COMM_RECEIVER_BUFSIZE]; + u8 *receiver_buffer; u8 serial; /* urb serial */ -- cgit v1.2.3-70-g09d2 From 45a14a8b50465a6ce61005f7fe9f3fd5c06823d5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Aug 2013 18:24:09 +0100 Subject: ASoC: ads711x: Add DAPM support This makes it easier to hook into boards and ensures the driver continues to work with support for non-DAPM CODECs removed. Signed-off-by: Mark Brown --- sound/soc/codecs/ads117x.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 506d474c4d22..8f388edff586 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -23,6 +23,28 @@ #define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000) #define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) +static const struct snd_soc_dapm_widget ads117x_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("Input1"), +SND_SOC_DAPM_INPUT("Input2"), +SND_SOC_DAPM_INPUT("Input3"), +SND_SOC_DAPM_INPUT("Input4"), +SND_SOC_DAPM_INPUT("Input5"), +SND_SOC_DAPM_INPUT("Input6"), +SND_SOC_DAPM_INPUT("Input7"), +SND_SOC_DAPM_INPUT("Input8"), +}; + +static const struct snd_soc_dapm_route ads117x_dapm_routes[] = { + { "Capture", NULL, "Input1" }, + { "Capture", NULL, "Input2" }, + { "Capture", NULL, "Input3" }, + { "Capture", NULL, "Input4" }, + { "Capture", NULL, "Input5" }, + { "Capture", NULL, "Input6" }, + { "Capture", NULL, "Input7" }, + { "Capture", NULL, "Input8" }, +}; + static struct snd_soc_dai_driver ads117x_dai = { /* ADC */ .name = "ads117x-hifi", @@ -34,7 +56,12 @@ static struct snd_soc_dai_driver ads117x_dai = { .formats = ADS117X_FORMATS,}, }; -static struct snd_soc_codec_driver soc_codec_dev_ads117x; +static struct snd_soc_codec_driver soc_codec_dev_ads117x = { + .dapm_widgets = ads117x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ads117x_dapm_widgets), + .dapm_routes = ads117x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ads117x_dapm_routes), +}; static int ads117x_probe(struct platform_device *pdev) { -- cgit v1.2.3-70-g09d2 From 57e6dae1087bbaa6b33d3dd8a8e90b63888939a3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 8 Aug 2013 11:24:55 +0200 Subject: ALSA: usb-audio: do not trust too-big wMaxPacketSize values The driver used to assume that the streaming endpoint's wMaxPacketSize value would be an indication of how much data the endpoint expects or sends, and compute the number of packets per URB using this value. However, the Focusrite Scarlett 2i4 declares a value of 1024 bytes, while only about 88 or 44 bytes are be actually used. This discrepancy would result in URBs with far too few packets, which would not work correctly on the EHCI driver. To get correct URBs, use wMaxPacketSize only as an upper limit on the packet size. Reported-by: James Stone Tested-by: James Stone Cc: # 2.6.35+ Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7a444b5501d9..659950e5b94f 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -591,17 +591,16 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; - /* calculate max. frequency */ - if (ep->maxpacksize) { + /* assume max. frequency is 25% higher than nominal */ + ep->freqmax = ep->freqn + (ep->freqn >> 2); + maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - ep->datainterval); + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ maxsize = ep->maxpacksize; ep->freqmax = (maxsize / (frame_bits >> 3)) << (16 - ep->datainterval); - } else { - /* no max. packet size: just take 25% higher than nominal */ - ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); } if (ep->fill_max) -- cgit v1.2.3-70-g09d2 From 16695971bec3b8b2398f7ab8dfa4c5a22bfcf95d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Aug 2013 12:25:57 +0100 Subject: ASoC: pcm1681: Staticise DAI driver It is not exported so doesn't need to be in the global namespace and sparse warns on this. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1681.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 27da41b2dfcd..51b18662e6aa 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -225,7 +225,7 @@ static const struct snd_kcontrol_new pcm1681_controls[] = { pcm1681_get_deemph, pcm1681_put_deemph), }; -struct snd_soc_dai_driver pcm1681_dai = { +static struct snd_soc_dai_driver pcm1681_dai = { .name = "pcm1681-hifi", .playback = { .stream_name = "Playback", -- cgit v1.2.3-70-g09d2 From 827d22f13618557bd35f938b020c954d83a82977 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Aug 2013 17:59:51 +0100 Subject: ASoC: ad73311: Add DAPM support This makes it possible to hook up other devices in boards and is required by removal of support for non-DAPM CODECs in the core. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/ad73311.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index b1f2baf42b48..5fac8adbc136 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -23,6 +23,21 @@ #include "ad73311.h" +static const struct snd_soc_dapm_widget ad73311_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINP"), +SND_SOC_DAPM_INPUT("VINN"), +SND_SOC_DAPM_OUTPUT("VOUTN"), +SND_SOC_DAPM_OUTPUT("VOUTP"), +}; + +static const struct snd_soc_dapm_route ad73311_dapm_routes[] = { + { "Capture", NULL, "VINP" }, + { "Capture", NULL, "VINN" }, + + { "VOUTN", NULL, "Playback" }, + { "VOUTP", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver ad73311_dai = { .name = "ad73311-hifi", .playback = { @@ -39,7 +54,12 @@ static struct snd_soc_dai_driver ad73311_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; -static struct snd_soc_codec_driver soc_codec_dev_ad73311; +static struct snd_soc_codec_driver soc_codec_dev_ad73311 = { + .dapm_widgets = ad73311_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad73311_dapm_widgets), + .dapm_routes = ad73311_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad73311_dapm_routes), +}; static int ad73311_probe(struct platform_device *pdev) { -- cgit v1.2.3-70-g09d2 From 74b45231b23d47b041b137737241d482481a76a9 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 8 Aug 2013 16:52:18 +0530 Subject: ASoC: s6105-ipcam: Fix incorrect placement of __initdata __initdata should be placed between the variable name and equal sign for the variable to be placed in the intended section. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/s6000/s6105-ipcam.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 58cfb1eb7dd3..945e8abdc10f 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -192,7 +192,7 @@ static struct snd_soc_card snd_soc_card_s6105 = { .num_links = 1, }; -static struct s6000_snd_platform_data __initdata s6105_snd_data = { +static struct s6000_snd_platform_data s6105_snd_data __initdata = { .wide = 0, .channel_in = 0, .channel_out = 1, -- cgit v1.2.3-70-g09d2 From 34d2f1b6feac3bc7e6022d30d624e9f3687717d3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 10 Aug 2013 09:53:14 +0200 Subject: ASoC: Remove unused soc_pm_waitq The soc_pm_waitq waitqueue has been around as long as the ASoC framework existed, but has never been used so far, so remove it. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2940e2c04525..c7d16df9efd9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -47,8 +47,6 @@ #define NAME_SIZE 32 -static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); - #ifdef CONFIG_DEBUG_FS struct dentry *snd_soc_debugfs_root; EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); -- cgit v1.2.3-70-g09d2 From 9c0aeaa3849150acaaf016202c6741d542b3c1df Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Sat, 10 Aug 2013 14:55:57 +0200 Subject: ASoC: imx-audmux: default configuration parser fixups Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 103d1b020496..ab17381cc981 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -278,7 +278,7 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, } for (i = 0; (ret = of_property_read_u32_index(child, - "fsl,port-config\n", i, &val)) == 0; + "fsl,port-config", i, &val)) == 0; ++i) { if (audmux_type == IMX31_AUDMUX) { if (i % 2) @@ -290,7 +290,7 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, } } - if (ret != -ENODATA) { + if (ret != -EOVERFLOW) { dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n", i, child->full_name); continue; -- cgit v1.2.3-70-g09d2 From c77f872e663e3f6ea18f774bf4399884884b4b22 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 10 Aug 2013 21:33:09 +0200 Subject: ASoC: Remove unused snd_soc_info_volsw_ext() The SOC_SINGLE_EXT control has been using snd_soc_info_volsw() for its info callback since commit 1c433fb ("[ALSA] soc - 0.13 ASoC headers"). The snd_soc_info_volsw_ext() function has been unused ever since then, so remove it. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 2 -- sound/soc/soc-core.c | 26 -------------------------- 2 files changed, 28 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 6eabee7ec15a..724a42af40fa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -497,8 +497,6 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); #define snd_soc_info_bool_ext snd_ctl_boolean_mono_info int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c7d16df9efd9..6ba5f7c23d3a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2577,32 +2577,6 @@ int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext); -/** - * snd_soc_info_volsw_ext - external single mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a single external mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int max = kcontrol->private_value; - - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = max; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); - /** * snd_soc_info_volsw - single mixer info callback * @kcontrol: mixer control -- cgit v1.2.3-70-g09d2 From 9a953e6f27fd280a2af5719b77394fbb228c5b46 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 10 Aug 2013 21:33:10 +0200 Subject: ASoC: Use snd_soc_info_enum_double() for SOC_ENUM_EXT controls snd_soc_info_enum_ext() and snd_soc_info_enum_double() are almost identical. The only difference is that snd_soc_info_enum_double() is also able to handle stereo controls. Using snd_soc_info_enum double() instead of snd_soc_info_enum_ext() for the SOC_ENUM_EXT control's info callback allows us to remove snd_soc_info_enum_ext(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 4 +--- sound/soc/soc-core.c | 27 --------------------------- 2 files changed, 1 insertion(+), 30 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 724a42af40fa..6f86a4187f58 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -234,7 +234,7 @@ .private_value = xdata } #define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_enum_ext, \ + .info = snd_soc_info_enum_double, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } @@ -485,8 +485,6 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6ba5f7c23d3a..f46472d50c9b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2550,33 +2550,6 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double); -/** - * snd_soc_info_enum_ext - external enumerated single mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about an external enumerated - * single mixer. - * - * Returns 0 for success. - */ -int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = e->max; - - if (uinfo->value.enumerated.item > e->max - 1) - uinfo->value.enumerated.item = e->max - 1; - strcpy(uinfo->value.enumerated.name, - e->texts[uinfo->value.enumerated.item]); - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext); - /** * snd_soc_info_volsw - single mixer info callback * @kcontrol: mixer control -- cgit v1.2.3-70-g09d2 From 439fe8a7bb07f8394fef03d7aa4f207166a32b88 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 13:26:43 +0100 Subject: ASoC: max9768: Add DAPM support This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown --- sound/soc/codecs/max9768.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index a6ac2313047d..31f91560e9f6 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -118,6 +118,18 @@ static const struct snd_kcontrol_new max9768_mute[] = { SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio), }; +static const struct snd_soc_dapm_widget max9768_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN"), + +SND_SOC_DAPM_OUTPUT("OUT+"), +SND_SOC_DAPM_OUTPUT("OUT-"), +}; + +static const struct snd_soc_dapm_route max9768_dapm_routes[] = { + { "OUT+", NULL, "IN" }, + { "OUT-", NULL, "IN" }, +}; + static int max9768_probe(struct snd_soc_codec *codec) { struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); @@ -148,6 +160,10 @@ static struct snd_soc_codec_driver max9768_codec_driver = { .probe = max9768_probe, .controls = max9768_volume, .num_controls = ARRAY_SIZE(max9768_volume), + .dapm_widgets = max9768_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9768_dapm_widgets), + .dapm_routes = max9768_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9768_dapm_routes), }; static const struct regmap_config max9768_i2c_regmap_config = { -- cgit v1.2.3-70-g09d2 From bad268f3504e2a58e406c3f0e282c1de629bd42f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 13:12:13 +0100 Subject: ASoC: cs4271: Convert to table based control init Signed-off-by: Mark Brown Acked-by: Alexander Sverdlin --- sound/soc/codecs/cs4271.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 03036b326732..65ad56c43c13 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -576,8 +576,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) CS4271_MODE2_MUTECAEQUB, CS4271_MODE2_MUTECAEQUB); - return snd_soc_add_codec_controls(codec, cs4271_snd_controls, - ARRAY_SIZE(cs4271_snd_controls)); + return 0; } static int cs4271_remove(struct snd_soc_codec *codec) @@ -596,6 +595,9 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, + + .controls = cs4271_snd_controls, + .num_controls = ARRAY_SIZE(cs4271_snd_controls), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From 2e7fb942a30563125d6aac497fa0dcddbb7d731d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 13:15:10 +0100 Subject: ASoC: cs4271: Add DAPM support This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown Acked-by: Alexander Sverdlin --- sound/soc/codecs/cs4271.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 65ad56c43c13..a20f1bb8f071 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -173,6 +173,26 @@ struct cs4271_private { bool enable_soft_reset; }; +static const struct snd_soc_dapm_widget cs4271_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINA"), +SND_SOC_DAPM_INPUT("AINB"), + +SND_SOC_DAPM_OUTPUT("AOUTA+"), +SND_SOC_DAPM_OUTPUT("AOUTA-"), +SND_SOC_DAPM_OUTPUT("AOUTB+"), +SND_SOC_DAPM_OUTPUT("AOUTB-"), +}; + +static const struct snd_soc_dapm_route cs4271_dapm_routes[] = { + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, + + { "AOUTA+", NULL, "Playback" }, + { "AOUTA-", NULL, "Playback" }, + { "AOUTB+", NULL, "Playback" }, + { "AOUTB-", NULL, "Playback" }, +}; + /* * @freq is the desired MCLK rate * MCLK rate should (c) be the sample rate, multiplied by one of the @@ -598,6 +618,10 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .controls = cs4271_snd_controls, .num_controls = ARRAY_SIZE(cs4271_snd_controls), + .dapm_widgets = cs4271_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4271_dapm_widgets), + .dapm_routes = cs4271_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From 4edec9eaf40877535b9b05cb0bf699f353c53418 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 11 Aug 2013 18:51:42 +0200 Subject: sound/soc/pxa/mioa701_wm9713.c: Avoid using ARRAY_AND_SIZE(e) as a function argument Replace ARRAY_AND_SIZE(e) in function argument position to avoid hiding the arity of the called function. The semantic match that makes this change is as follows: (http://coccinelle.lip6.fr/) // @@ expression e,f; @@ f(..., - ARRAY_AND_SIZE(e) + e,ARRAY_SIZE(e) ,...) // Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/pxa/mioa701_wm9713.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 97b711e12821..20fdce61060c 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -133,10 +133,11 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) unsigned short reg; /* Add mioa701 specific widgets */ - snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets, + ARRAY_SIZE(mioa701_dapm_widgets)); /* Set up mioa701 specific audio path audio_mapnects */ - snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Prepare GPIO8 for rear speaker amplifier */ reg = codec->driver->read(codec, AC97_GPIO_CFG); -- cgit v1.2.3-70-g09d2 From db8a38e5063a4daf61252e65d47ab3495c705f4c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Aug 2013 12:34:42 +0200 Subject: ALSA: hda - Add pinfix for LG LW25 laptop Correct the pins for a line-in and a headphone on LG LW25 laptop with ALC880 codec. Other pins seem fine. Reported-and-tested-by: Joonas Saarinen Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd226149868..5b22bf958764 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1031,6 +1031,7 @@ enum { ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, ALC880_FIXUP_LG, + ALC880_FIXUP_LG_LW25, ALC880_FIXUP_W810, ALC880_FIXUP_EAPD_COEF, ALC880_FIXUP_TCL_S700, @@ -1089,6 +1090,14 @@ static const struct hda_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_LG_LW25] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x0181344f }, /* line-in */ + { 0x1b, 0x0321403f }, /* headphone */ + { } + } + }, [ALC880_FIXUP_W810] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -1341,6 +1350,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), + SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_FIXUP_LG_LW25), SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700), /* Below is the copied entries from alc880_quirks.c. -- cgit v1.2.3-70-g09d2 From 5ece263f1d93fba8d992e67e3ab8a71acf674db9 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Sun, 11 Aug 2013 11:11:19 +0200 Subject: ALSA: 6fire: make buffers DMA-able (pcm) Patch makes pcm buffers DMA-able by allocating each one separately. Signed-off-by: Torsten Schenk Cc: Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 41 ++++++++++++++++++++++++++++++++++++++++- sound/usb/6fire/pcm.h | 2 +- 2 files changed, 41 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 3d2551cc10f2..b5eb97fdc842 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -582,6 +582,33 @@ static void usb6fire_pcm_init_urb(struct pcm_urb *urb, urb->instance.number_of_packets = PCM_N_PACKETS_PER_URB; } +static int usb6fire_pcm_buffers_init(struct pcm_runtime *rt) +{ + int i; + + for (i = 0; i < PCM_N_URBS; i++) { + rt->out_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB + * PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!rt->out_urbs[i].buffer) + return -ENOMEM; + rt->in_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB + * PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!rt->in_urbs[i].buffer) + return -ENOMEM; + } + return 0; +} + +static void usb6fire_pcm_buffers_destroy(struct pcm_runtime *rt) +{ + int i; + + for (i = 0; i < PCM_N_URBS; i++) { + kfree(rt->out_urbs[i].buffer); + kfree(rt->in_urbs[i].buffer); + } +} + int usb6fire_pcm_init(struct sfire_chip *chip) { int i; @@ -593,6 +620,13 @@ int usb6fire_pcm_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + ret = usb6fire_pcm_buffers_init(rt); + if (ret) { + usb6fire_pcm_buffers_destroy(rt); + kfree(rt); + return ret; + } + rt->chip = chip; rt->stream_state = STREAM_DISABLED; rt->rate = ARRAY_SIZE(rates); @@ -614,6 +648,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip) ret = snd_pcm_new(chip->card, "DMX6FireUSB", 0, 1, 1, &pcm); if (ret < 0) { + usb6fire_pcm_buffers_destroy(rt); kfree(rt); snd_printk(KERN_ERR PREFIX "cannot create pcm instance.\n"); return ret; @@ -625,6 +660,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_ops); if (ret) { + usb6fire_pcm_buffers_destroy(rt); kfree(rt); snd_printk(KERN_ERR PREFIX "error preallocating pcm buffers.\n"); @@ -669,6 +705,9 @@ void usb6fire_pcm_abort(struct sfire_chip *chip) void usb6fire_pcm_destroy(struct sfire_chip *chip) { - kfree(chip->pcm); + struct pcm_runtime *rt = chip->pcm; + + usb6fire_pcm_buffers_destroy(rt); + kfree(rt); chip->pcm = NULL; } diff --git a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h index 9b01133ee3fe..f5779d6182c6 100644 --- a/sound/usb/6fire/pcm.h +++ b/sound/usb/6fire/pcm.h @@ -32,7 +32,7 @@ struct pcm_urb { struct urb instance; struct usb_iso_packet_descriptor packets[PCM_N_PACKETS_PER_URB]; /* END DO NOT SEPARATE */ - u8 buffer[PCM_N_PACKETS_PER_URB * PCM_MAX_PACKET_SIZE]; + u8 *buffer; struct pcm_urb *peer; }; -- cgit v1.2.3-70-g09d2 From 4c2aee0032b70083dafebd733ed9c774633b2fa3 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Sun, 11 Aug 2013 11:11:35 +0200 Subject: ALSA: 6fire: make buffers DMA-able (midi) Patch makes midi output buffer DMA-able by allocating it separately. Signed-off-by: Torsten Schenk Cc: Signed-off-by: Takashi Iwai --- sound/usb/6fire/midi.c | 16 +++++++++++++++- sound/usb/6fire/midi.h | 6 +----- 2 files changed, 16 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c index 26722423330d..f3dd7266c391 100644 --- a/sound/usb/6fire/midi.c +++ b/sound/usb/6fire/midi.c @@ -19,6 +19,10 @@ #include "chip.h" #include "comm.h" +enum { + MIDI_BUFSIZE = 64 +}; + static void usb6fire_midi_out_handler(struct urb *urb) { struct midi_runtime *rt = urb->context; @@ -156,6 +160,12 @@ int usb6fire_midi_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL); + if (!rt->out_buffer) { + kfree(rt); + return -ENOMEM; + } + rt->chip = chip; rt->in_received = usb6fire_midi_in_received; rt->out_buffer[0] = 0x80; /* 'send midi' command */ @@ -169,6 +179,7 @@ int usb6fire_midi_init(struct sfire_chip *chip) ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance); if (ret < 0) { + kfree(rt->out_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "unable to create midi.\n"); return ret; @@ -197,6 +208,9 @@ void usb6fire_midi_abort(struct sfire_chip *chip) void usb6fire_midi_destroy(struct sfire_chip *chip) { - kfree(chip->midi); + struct midi_runtime *rt = chip->midi; + + kfree(rt->out_buffer); + kfree(rt); chip->midi = NULL; } diff --git a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h index c321006e5430..84851b9f5559 100644 --- a/sound/usb/6fire/midi.h +++ b/sound/usb/6fire/midi.h @@ -16,10 +16,6 @@ #include "common.h" -enum { - MIDI_BUFSIZE = 64 -}; - struct midi_runtime { struct sfire_chip *chip; struct snd_rawmidi *instance; @@ -32,7 +28,7 @@ struct midi_runtime { struct snd_rawmidi_substream *out; struct urb out_urb; u8 out_serial; /* serial number of out packet */ - u8 out_buffer[MIDI_BUFSIZE]; + u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length); -- cgit v1.2.3-70-g09d2 From aa773bfe8f860173752258c9ba4bf51060fb0d07 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 11 Aug 2013 14:13:13 +0200 Subject: ALSA: usb-audio: fix automatic Roland/Yamaha MIDI detection Commit aafe77cc45a5 (ALSA: usb-audio: add support for many Roland/Yamaha devices) had several logic errors that prevented create_auto_midi_quirk from enumerating any MIDI ports. Reported-by: Keith A. Milner Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 1bc45e71f1fe..0df9ede99dfd 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -319,19 +319,19 @@ static int create_auto_midi_quirk(struct snd_usb_audio *chip, if (altsd->bNumEndpoints < 1) return -ENODEV; epd = get_endpoint(alts, 0); - if (!usb_endpoint_xfer_bulk(epd) || + if (!usb_endpoint_xfer_bulk(epd) && !usb_endpoint_xfer_int(epd)) return -ENODEV; switch (USB_ID_VENDOR(chip->usb_id)) { case 0x0499: /* Yamaha */ err = create_yamaha_midi_quirk(chip, iface, driver, alts); - if (err < 0 && err != -ENODEV) + if (err != -ENODEV) return err; break; case 0x0582: /* Roland */ err = create_roland_midi_quirk(chip, iface, driver, alts); - if (err < 0 && err != -ENODEV) + if (err != -ENODEV) return err; break; } -- cgit v1.2.3-70-g09d2 From f69910ddbd8c29391958cf82b598dd78fe5c8640 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Aug 2013 09:32:37 +0200 Subject: ALSA: hda - Fix missing mute controls for CX5051 We've added a fake mute control (setting the amp volume to zero) for CX5051 at commit [3868137e: ALSA: hda - Add a fake mute feature], but this feature was overlooked in the generic parser implementation. Now the driver lacks of mute controls on these codecs. The fix is just to check both AC_AMPCAP_MUTE and AC_AMPCAP_MIN_MUTE bits in each place checking the amp capabilities. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59001 Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8e77cbbad871..e3c7ba8d7582 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -522,7 +522,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1, } #define nid_has_mute(codec, nid, dir) \ - check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE) + check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) #define nid_has_volume(codec, nid, dir) \ check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS) @@ -624,7 +624,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, if (enable) val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; } - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (!enable) val |= HDA_AMP_MUTE; } @@ -648,7 +648,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec, { unsigned int mask = 0xff; - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL)) mask &= ~0x80; } -- cgit v1.2.3-70-g09d2 From 3d24cfe485e2750cc209a77dd62fa1fe004fc6c7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 9 Aug 2013 18:12:29 +0100 Subject: ASoC: compress: Use power efficient workqueue There is no need for the power down work to be done on a per CPU workqueue especially considering the fairly long delay before powerdown. Signed-off-by: Mark Brown Acked-by: Vinod Koul --- sound/soc/soc-compress.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index d22015074670..53c9ecdd119f 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -149,8 +149,9 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SND_SOC_DAPM_STREAM_STOP); } else { rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ -- cgit v1.2.3-70-g09d2 From 9190aeb4ecbdcab7d66d186c207f76d09b41d082 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 15:07:36 +0100 Subject: ASoC: adau1701: Use gpio_set_value_cansleep() The GPIO manipulation done by this driver is never in atomic context so we can use gpio_set_value_cansleep() and support GPIOs that can't be set from atomic context. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/adau1701.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 2c102522bbbc..ebff1128be59 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -247,21 +247,21 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) gpio_is_valid(adau1701->gpio_pll_mode[1])) { switch (clkdiv) { case 64: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); break; case 256: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; case 384: - gpio_set_value(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); break; case 0: /* fallback */ case 512: - gpio_set_value(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; } } @@ -269,10 +269,10 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) adau1701->pll_clkdiv = clkdiv; if (gpio_is_valid(adau1701->gpio_nreset)) { - gpio_set_value(adau1701->gpio_nreset, 0); + gpio_set_value_cansleep(adau1701->gpio_nreset, 0); /* minimum reset time is 20ns */ udelay(1); - gpio_set_value(adau1701->gpio_nreset, 1); + gpio_set_value_cansleep(adau1701->gpio_nreset, 1); /* power-up time may be as long as 85ms */ mdelay(85); } -- cgit v1.2.3-70-g09d2 From 2e61926cb4a42d79a406aa64f04869d1227ca42c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Aug 2013 19:01:40 +0100 Subject: ASoC: ak4104: Add stub DAPM support This makes it easer to integrate the device with other on-board components and ensures correct operation following removal of support for non-DAPM CODECs. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index c7cfdf957e4d..9a7c89b9cb35 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -51,6 +51,14 @@ struct ak4104_private { struct regmap *regmap; }; +static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ak4104_dapm_routes[] = { + { "TX", NULL, "Playback" }, +}; + static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { @@ -214,6 +222,11 @@ static int ak4104_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, + + .dapm_widgets = ak4104_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4104_dapm_widgets), + .dapm_routes = ak4104_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4104_dapm_routes), }; static const struct regmap_config ak4104_regmap = { -- cgit v1.2.3-70-g09d2 From a5db4d50fa578936275c1e26d5d2fda25c0d2bf6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Aug 2013 19:05:47 +0100 Subject: ASoC: ak4104: Manage TXE using DAPM Saves some code. We should also be able to manage the power up and reset registers using DAPM but it's probably more trouble than it's worth in mains powered systems. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 23 ++++------------------- 1 file changed, 4 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 9a7c89b9cb35..71059c07ae7b 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -52,11 +52,14 @@ struct ak4104_private { }; static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = { +SND_SOC_DAPM_PGA("TXE", AK4104_REG_TX, AK4104_TX_TXE, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("TX"), }; static const struct snd_soc_dapm_route ak4104_dapm_routes[] = { - { "TX", NULL, "Playback" }, + { "TXE", NULL, "Playback" }, + { "TX", NULL, "TXE" }, }; static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, @@ -146,29 +149,11 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* enable transmitter */ - ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, - AK4104_TX_TXE, AK4104_TX_TXE); - if (ret < 0) - return ret; - return 0; } -static int ak4104_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); - - /* disable transmitter */ - return regmap_update_bits(ak4104->regmap, AK4104_REG_TX, - AK4104_TX_TXE, 0); -} - static const struct snd_soc_dai_ops ak4101_dai_ops = { .hw_params = ak4104_hw_params, - .hw_free = ak4104_hw_free, .set_fmt = ak4104_set_dai_fmt, }; -- cgit v1.2.3-70-g09d2 From dcf1439a493f75336f7e9d272d01b04bc1c4ca8e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 12:28:56 +0100 Subject: ASoC: ak5386: Add DAPM support This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown --- sound/soc/codecs/ak5386.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 1f303983ae02..72e953b2cb41 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -22,7 +22,22 @@ struct ak5386_priv { int reset_gpio; }; -static struct snd_soc_codec_driver soc_codec_ak5386; +static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route ak5386_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + +static struct snd_soc_codec_driver soc_codec_ak5386 = { + .dapm_widgets = ak5386_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets), + .dapm_routes = ak5386_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak5386_dapm_routes), +}; static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) -- cgit v1.2.3-70-g09d2 From e7edb2731bf8e00aaeb7d20800ae108068618f63 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 12 Aug 2013 11:33:32 +0100 Subject: ASoC: arizona: Add widget<->mux route into mux route macro The routes linking the widget and the input mux were being added manually, rather than by the ARIZONA_MUX_ROUTES macro. This patchs adds the routes to the macro. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.h | 3 ++- sound/soc/codecs/wm5102.c | 41 ++++++++++++----------------------------- sound/soc/codecs/wm5110.c | 8 ++++---- 3 files changed, 18 insertions(+), 34 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b6b6d7036ea0..9e81b6392692 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -150,7 +150,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MUX(name_str " Aux 5", &name##_aux5_mux), \ ARIZONA_MUX(name_str " Aux 6", &name##_aux6_mux) -#define ARIZONA_MUX_ROUTES(name) \ +#define ARIZONA_MUX_ROUTES(widget, name) \ + { widget, NULL, name " Input" }, \ ARIZONA_MIXER_INPUT_ROUTES(name " Input") #define ARIZONA_MIXER_ROUTES(widget, name) \ diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a6cbdb4b5c0f..f38c52d43b8d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1501,23 +1501,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "IN3L PGA", NULL, "IN3L" }, { "IN3R PGA", NULL, "IN3R" }, - { "ASRC1L", NULL, "ASRC1L Input" }, - { "ASRC1R", NULL, "ASRC1R Input" }, - { "ASRC2L", NULL, "ASRC2L Input" }, - { "ASRC2R", NULL, "ASRC2R Input" }, - - { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" }, - { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" }, - - { "ISRC1INT1", NULL, "ISRC1INT1 Input" }, - { "ISRC1INT2", NULL, "ISRC1INT2 Input" }, - - { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" }, - { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" }, - - { "ISRC2INT1", NULL, "ISRC2INT1 Input" }, - { "ISRC2INT2", NULL, "ISRC2INT2 Input" }, - ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), @@ -1569,22 +1552,22 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), - ARIZONA_MUX_ROUTES("ISRC1INT1"), - ARIZONA_MUX_ROUTES("ISRC1INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), - ARIZONA_MUX_ROUTES("ISRC1DEC1"), - ARIZONA_MUX_ROUTES("ISRC1DEC2"), + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), - ARIZONA_MUX_ROUTES("ISRC2INT1"), - ARIZONA_MUX_ROUTES("ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), - ARIZONA_MUX_ROUTES("ISRC2DEC1"), - ARIZONA_MUX_ROUTES("ISRC2DEC2"), + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), ARIZONA_DSP_ROUTES("DSP1"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 77fd531bf3cc..38e50c81a953 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -978,10 +978,10 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, -- cgit v1.2.3-70-g09d2 From 140d37de62ffe8405282a1d6498f3b4099006384 Mon Sep 17 00:00:00 2001 From: "Maksim A. Boyko" Date: Sat, 10 Aug 2013 12:20:02 +0400 Subject: ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525 Add the volume control quirk for avoiding the kernel warning for the Logitech HD Webcam C525 as in the similar commit 36691e1be6ec551eef4a5225f126a281f8c051c2 for the Logitech HD Webcam C310. Reported-by: Maksim Boyko Tested-by: Maksim Boyko Cc: # 3.10.5+ Signed-off-by: Maksim Boyko Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d5438083fd6a..95558ef4a7a0 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -888,6 +888,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. -- cgit v1.2.3-70-g09d2 From 946d92a100f6c36b1c53922d5105b3c19a59173d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Aug 2013 23:28:42 +0100 Subject: ASoC: dapm: Don't create routes when creating kcontrols Attempting to create the route as part of adding a mux control causes us to attempt to add the same route twice since we loop over all sources for the mux after creating the control. Instead do the addition in the callers. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/soc-dapm.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 662a904c2b79..b885a9bedc4e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -665,7 +665,7 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, * create it. Either way, add the widget into the control's widget list */ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, - int kci, struct snd_soc_dapm_path *path) + int kci) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; @@ -766,7 +766,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, return ret; w->kcontrols[kci] = kcontrol; - dapm_kcontrol_add_path(kcontrol, path); return 0; } @@ -790,9 +789,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) continue; } - ret = dapm_create_or_share_mixmux_kcontrol(w, i, path); + ret = dapm_create_or_share_mixmux_kcontrol(w, i); if (ret < 0) return ret; + + dapm_kcontrol_add_path(w->kcontrols[i], path); } } @@ -818,10 +819,7 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - path = list_first_entry(&w->sources, struct snd_soc_dapm_path, - list_sink); - - ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From 69c2d346e8fa8dbed122e82f727332f35718ab86 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 00:20:36 +0100 Subject: ASoC: dapm: Ensure kcontrol list is initialised Ensure that the recently added path kcontrol list is initialised otherwise we may crash trying to delete routes that don't have kcontrols. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b885a9bedc4e..d84bd0f167b6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2354,6 +2354,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, path->sink = wsink; path->connected = connected; INIT_LIST_HEAD(&path->list); + INIT_LIST_HEAD(&path->list_kcontrol); INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); -- cgit v1.2.3-70-g09d2 From 40843aea5a9bd2c3d7917d086e6d23cb02cc4b39 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 12 Aug 2013 23:46:55 +0100 Subject: ASoC: wm8997: Initial CODEC driver The wm8997 is a compact, high-performance audio hub CODEC with SLIMbus interfacing, for smartphones, tablets and other portable audio devices based on the Arizona platform. This patch adds the wm8997 CODEC driver. [Fixed some interface churn from bitrot due to the patch not going via the MFD tree as expected -- broonie] Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 13 +- sound/soc/codecs/wm8997.c | 1175 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8997.h | 23 + 5 files changed, 1216 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/wm8997.c create mode 100644 sound/soc/codecs/wm8997.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..bb34c8a4bf0e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8996 if I2C + select SND_SOC_WM8997 if MFD_WM8997 select SND_SOC_WM9081 if I2C select SND_SOC_WM9090 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS @@ -145,8 +146,10 @@ config SND_SOC_ARIZONA tristate default y if SND_SOC_WM5102=y default y if SND_SOC_WM5110=y + default y if SND_SOC_WM8997=y default m if SND_SOC_WM5102=m default m if SND_SOC_WM5110=m + default m if SND_SOC_WM8997=m config SND_SOC_WM_HUBS tristate @@ -500,6 +503,9 @@ config SND_SOC_WM8995 config SND_SOC_WM8996 tristate +config SND_SOC_WM8997 + tristate + config SND_SOC_WM9081 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd8066f546..68ea0a2c169c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -114,6 +114,7 @@ snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o snd-soc-wm8995-objs := wm8995.o +snd-soc-wm8997-objs := wm8997.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9090-objs := wm9090.o snd-soc-wm9705-objs := wm9705.o @@ -239,6 +240,7 @@ obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o +obj-$(CONFIG_SND_SOC_WM8997) += snd-soc-wm8997.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 779a0eeac67c..657808ba1418 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -200,9 +200,16 @@ int arizona_init_spk(struct snd_soc_codec *codec) if (ret != 0) return ret; - ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1); - if (ret != 0) - return ret; + switch (arizona->type) { + case WM8997: + break; + default: + ret = snd_soc_dapm_new_controls(&codec->dapm, + &arizona_spkr, 1); + if (ret != 0) + return ret; + break; + } ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN, "Thermal warning", arizona_thermal_warn, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c new file mode 100644 index 000000000000..0a43bac2f4e3 --- /dev/null +++ b/sound/soc/codecs/wm8997.c @@ -0,0 +1,1175 @@ +/* + * wm8997.c -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Charles Keepax + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include "arizona.h" +#include "wm8997.h" + +struct wm8997_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); + +static const struct reg_default wm8997_sysclk_reva_patch[] = { + { 0x301D, 0x7B15 }, + { 0x301B, 0x0050 }, + { 0x305D, 0x7B17 }, + { 0x305B, 0x0050 }, + { 0x3001, 0x08FE }, + { 0x3003, 0x00F4 }, + { 0x3041, 0x08FF }, + { 0x3043, 0x0005 }, + { 0x3020, 0x0225 }, + { 0x3021, 0x0A00 }, + { 0x3022, 0xE24D }, + { 0x3023, 0x0800 }, + { 0x3024, 0xE24D }, + { 0x3025, 0xF000 }, + { 0x3060, 0x0226 }, + { 0x3061, 0x0A00 }, + { 0x3062, 0xE252 }, + { 0x3063, 0x0800 }, + { 0x3064, 0xE252 }, + { 0x3065, 0xF000 }, + { 0x3116, 0x022B }, + { 0x3117, 0xFA00 }, + { 0x3110, 0x246C }, + { 0x3111, 0x0A03 }, + { 0x3112, 0x246E }, + { 0x3113, 0x0A03 }, + { 0x3114, 0x2470 }, + { 0x3115, 0x0A03 }, + { 0x3126, 0x246C }, + { 0x3127, 0x0A02 }, + { 0x3128, 0x246E }, + { 0x3129, 0x0A02 }, + { 0x312A, 0x2470 }, + { 0x312B, 0xFA02 }, + { 0x3125, 0x0800 }, +}; + +static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 0: + patch = wm8997_sysclk_reva_patch; + patch_size = ARRAY_SIZE(wm8997_sysclk_reva_patch); + break; + default: + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + default: + break; + } + + return 0; +} + +static const char *wm8997_osr_text[] = { + "Low power", "Normal", "High performance", +}; + +static const unsigned int wm8997_osr_val[] = { + 0x0, 0x3, 0x5, +}; + +static const struct soc_enum wm8997_hpout_osr[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), +}; + +#define WM8997_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG EPOUT Switch", base, 4, 1, 0), \ + SOC_SINGLE(name " NG SPKOUT Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0) + +static const struct snd_kcontrol_new wm8997_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), + +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), + +SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), +SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21, + ARIZONA_EQ1_ENA_MASK), +SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21, + ARIZONA_EQ2_ENA_MASK), +SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21, + ARIZONA_EQ3_ENA_MASK), +SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21, + ARIZONA_EQ4_ENA_MASK), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), +SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), + +SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUT", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_VALUE_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]), +SOC_VALUE_ENUM("EPOUT OSR", wm8997_hpout_osr[1]), + +SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), +SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), +SOC_ENUM("Noise Gate Hold", arizona_ng_hold), + +WM8997_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), +WM8997_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), +WM8997_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L), +WM8997_NG_SRC("SPKOUT", ARIZONA_NOISE_GATE_SELECT_4L), +WM8997_NG_SRC("SPKDAT1L", ARIZONA_NOISE_GATE_SELECT_5L), +WM8997_NG_SRC("SPKDAT1R", ARIZONA_NOISE_GATE_SELECT_5R), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUT, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); + +static const char *wm8997_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "EPOUT", "SPKOUT", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int wm8997_aec_loopback_values[] = { + 0, 1, 4, 6, 8, 9, +}; + +static const struct soc_enum wm8997_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(wm8997_aec_loopback_texts), + wm8997_aec_loopback_texts, + wm8997_aec_loopback_values); + +static const struct snd_kcontrol_new wm8997_aec_loopback_mux = + SOC_DAPM_VALUE_ENUM("AEC Loopback", wm8997_aec_loopback); + +static const struct snd_soc_dapm_widget wm8997_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, wm8997_sysclk_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), +SND_SOC_DAPM_SIGGEN("HAPTICS"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8997_aec_loopback_mux), + +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUT, "SPKOUT"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"), +ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"), +ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"), + +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), + +SND_SOC_DAPM_OUTPUT("MICSUPP"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "Haptics", "HAPTICS" }, \ + { name, "AEC", "AEC Loopback" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "SLIMRX5", "SLIMRX5" }, \ + { name, "SLIMRX6", "SLIMRX6" }, \ + { name, "SLIMRX7", "SLIMRX7" }, \ + { name, "SLIMRX8", "SLIMRX8" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" } + +static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDD" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + { "SLIMRX3", NULL, "Slim1 Playback" }, + { "SLIMRX4", NULL, "Slim1 Playback" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX5", NULL, "Slim2 Playback" }, + { "SLIMRX6", NULL, "Slim2 Playback" }, + + { "Slim3 Capture", NULL, "SLIMTX7" }, + { "Slim3 Capture", NULL, "SLIMTX8" }, + + { "SLIMRX7", NULL, "Slim3 Playback" }, + { "SLIMRX8", NULL, "Slim3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + { "Slim3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + { "Slim3 Capture", NULL, "SYSCLK" }, + + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUT"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"), + ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"), + ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), + + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), + + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "AEC Loopback", "EPOUT", "OUT3L" }, + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "AEC Loopback", "SPKOUT", "OUT4L" }, + { "SPKOUTN", NULL, "OUT4L" }, + { "SPKOUTP", NULL, "OUT4L" }, + + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "MICSUPP", NULL, "SYSCLK" }, +}; + +static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm8997_priv *wm8997 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM8997_FLL1: + return arizona_set_fll(&wm8997->fll[0], source, Fref, Fout); + case WM8997_FLL2: + return arizona_set_fll(&wm8997->fll[1], source, Fref, Fout); + case WM8997_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[0], source, Fref, + Fout); + case WM8997_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[1], source, Fref, + Fout); + default: + return -EINVAL; + } +} + +#define WM8997_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm8997_dai[] = { + { + .name = "wm8997-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-slim1", + .id = 3, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim2", + .id = 4, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim3", + .id = 5, + .playback = { + .stream_name = "Slim3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, +}; + +static int wm8997_codec_probe(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = priv->core.arizona->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + if (ret != 0) + return ret; + + arizona_init_spk(codec); + + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); + + priv->core.arizona->dapm = &codec->dapm; + + return 0; +} + +static int wm8997_codec_remove(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->core.arizona->dapm = NULL; + + return 0; +} + +#define WM8997_DIG_VU 0x0200 + +static unsigned int wm8997_digital_vu[] = { + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8997 = { + .probe = wm8997_codec_probe, + .remove = wm8997_codec_remove, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm8997_set_fll, + + .controls = wm8997_snd_controls, + .num_controls = ARRAY_SIZE(wm8997_snd_controls), + .dapm_widgets = wm8997_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8997_dapm_widgets), + .dapm_routes = wm8997_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8997_dapm_routes), +}; + +static int wm8997_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm8997_priv *wm8997; + int i; + + wm8997 = devm_kzalloc(&pdev->dev, sizeof(struct wm8997_priv), + GFP_KERNEL); + if (wm8997 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm8997); + + wm8997->core.arizona = arizona; + wm8997->core.num_inputs = 4; + + for (i = 0; i < ARRAY_SIZE(wm8997->fll); i++) + wm8997->fll[i].vco_mult = 1; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm8997->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm8997->fll[1]); + + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + + for (i = 0; i < ARRAY_SIZE(wm8997_dai); i++) + arizona_init_dai(&wm8997->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm8997_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm8997_digital_vu[i], + WM8997_DIG_VU, WM8997_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8997, + wm8997_dai, ARRAY_SIZE(wm8997_dai)); +} + +static int wm8997_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm8997_codec_driver = { + .driver = { + .name = "wm8997-codec", + .owner = THIS_MODULE, + }, + .probe = wm8997_probe, + .remove = wm8997_remove, +}; + +module_platform_driver(wm8997_codec_driver); + +MODULE_DESCRIPTION("ASoC WM8997 driver"); +MODULE_AUTHOR("Charles Keepax "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8997-codec"); diff --git a/sound/soc/codecs/wm8997.h b/sound/soc/codecs/wm8997.h new file mode 100644 index 000000000000..5e91c6a7d567 --- /dev/null +++ b/sound/soc/codecs/wm8997.h @@ -0,0 +1,23 @@ +/* + * wm8997.h -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8997_H +#define _WM8997_H + +#include "arizona.h" + +#define WM8997_FLL1 1 +#define WM8997_FLL2 2 +#define WM8997_FLL1_REFCLK 3 +#define WM8997_FLL2_REFCLK 4 + +#endif -- cgit v1.2.3-70-g09d2 From 4601736a6f8e7ae09f1010df02e1ced605043cad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 12:28:42 +0100 Subject: ASoC: ak4554: Add DAPM support This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4554.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index 6aed9c4d06d6..79e9555766c0 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -29,6 +29,22 @@ * CPU-DAI2 (capture only fmt = LEFT_J) ---+ */ +static const struct snd_soc_dapm_widget ak4554_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), + +SND_SOC_DAPM_OUTPUT("AOUTL"), +SND_SOC_DAPM_OUTPUT("AOUTR"), +}; + +static const struct snd_soc_dapm_route ak4554_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, + + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver ak4554_dai = { .name = "ak4554-hifi", .playback = { @@ -49,6 +65,10 @@ static struct snd_soc_dai_driver ak4554_dai = { }; static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { + .dapm_widgets = ak4554_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4554_dapm_widgets), + .dapm_routes = ak4554_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4554_dapm_routes), }; static int ak4554_soc_probe(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From 997288e3824e4c6a7ab4ca4f580fe35e138d62e8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 13:10:19 +0100 Subject: ASoC: max9877: Convert to use regmap API Signed-off-by: Mark Brown --- sound/soc/codecs/max9877.c | 187 ++++++++++++++++++++++++++++----------------- 1 file changed, 118 insertions(+), 69 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 6b6c74cd83e2..7e2fe5023fb2 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -14,27 +14,21 @@ #include #include #include +#include #include #include #include "max9877.h" -static struct i2c_client *i2c; +static struct regmap *regmap; -static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; - -static void max9877_write_regs(void) -{ - unsigned int i; - u8 data[6]; - - data[0] = MAX9877_INPUT_MODE; - for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) - data[i + 1] = max9877_regs[i]; - - if (i2c_master_send(i2c, data, 6) != 6) - dev_err(&i2c->dev, "i2c write failed\n"); -} +static struct reg_default max9877_regs[] = { + { 0, 0x40 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x49 }, +}; static int max9877_get_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -45,8 +39,14 @@ static int max9877_get_reg(struct snd_kcontrol *kcontrol, unsigned int shift = mc->shift; unsigned int mask = mc->max; unsigned int invert = mc->invert; + unsigned int val; + int ret; + + ret = regmap_read(regmap, reg, &val); + if (ret != 0) + return ret; - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = @@ -65,18 +65,21 @@ static int max9877_set_reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int invert = mc->invert; unsigned int val = (ucontrol->value.integer.value[0] & mask); + bool change; + int ret; if (invert) val = mask - val; - if (((max9877_regs[reg] >> shift) & mask) == val) - return 0; - - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_write_regs(); + ret = regmap_update_bits_check(regmap, reg, mask << shift, + val << shift, &change); + if (ret != 0) + return ret; - return 1; + if (change) + return 1; + else + return 0; } static int max9877_get_2reg(struct snd_kcontrol *kcontrol, @@ -88,9 +91,18 @@ static int max9877_get_2reg(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; unsigned int mask = mc->max; + unsigned int val; + int ret; + + ret = regmap_read(regmap, reg, &val); + if (ret != 0) + return ret; + ucontrol->value.integer.value[0] = (val >> shift) & mask; - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; - ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask; + ret = regmap_read(regmap, reg2, &val); + if (ret != 0) + return ret; + ucontrol->value.integer.value[1] = (val >> shift) & mask; return 0; } @@ -106,77 +118,99 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int val = (ucontrol->value.integer.value[0] & mask); unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - unsigned int change = 0; - - if (((max9877_regs[reg] >> shift) & mask) != val) - change = 1; - - if (((max9877_regs[reg2] >> shift) & mask) != val2) - change = 1; - - if (change) { - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_regs[reg2] &= ~(mask << shift); - max9877_regs[reg2] |= val2 << shift; - max9877_write_regs(); - } - - return change; + bool change1, change2; + int ret; + + ret = regmap_update_bits_check(regmap, reg, mask << shift, + val << shift, &change1); + if (ret != 0) + return ret; + + ret = regmap_update_bits_check(regmap, reg2, mask << shift, + val2 << shift, &change2); + if (ret != 0) + return ret; + + if (change1 || change2) + return 1; + else + return 0; } static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; + unsigned int val; + int ret; + + ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); + if (ret != 0) + return ret; + + val &= MAX9877_OUTMODE_MASK; + if (val) + val--; - if (value) - value -= 1; + ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[0] = value; return 0; } static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = ucontrol->value.integer.value[0]; + unsigned int val; + bool change; + int ret; - value += 1; + val = ucontrol->value.integer.value[0] + 1; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) - return 0; + ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, + MAX9877_OUTMODE_MASK, val, &change); + if (ret != 0) + return ret; - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; + if (change) + return 1; + else + return 0; } static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); + unsigned int val; + int ret; + + ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); + if (ret != 0) + return ret; + + val &= MAX9877_OSC_MASK; + val >>= MAX9877_OSC_OFFSET; - value = value >> MAX9877_OSC_OFFSET; + ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[0] = value; return 0; } static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = ucontrol->value.integer.value[0]; - - value = value << MAX9877_OSC_OFFSET; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) + unsigned int val; + bool change; + int ret; + + val = ucontrol->value.integer.value[0] << MAX9877_OSC_OFFSET; + ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, + MAX9877_OSC_MASK, val, &change); + if (ret != 0) + return ret; + + if (change) + return 1; + else return 0; - - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; } static const unsigned int max9877_pgain_tlv[] = { @@ -258,19 +292,34 @@ int max9877_add_controls(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(max9877_add_controls); +static const struct regmap_config max9877_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = max9877_regs, + .num_reg_defaults = ARRAY_SIZE(max9877_regs), + .cache_type = REGCACHE_RBTREE, +}; + static int max9877_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - i2c = client; + int i; + + regmap = devm_regmap_init_i2c(client, &max9877_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - max9877_write_regs(); + /* Ensure the device is in reset state */ + for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) + regmap_write(regmap, max9877_regs[i].reg, max9877_regs[i].def); return 0; } static int max9877_i2c_remove(struct i2c_client *client) { - i2c = NULL; + regmap = NULL; return 0; } -- cgit v1.2.3-70-g09d2 From d76a96174b31bd916c1dfaa81a3db82fc8c54b91 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 13:20:15 +0100 Subject: ASoC: max9877: Convert to standard CODEC driver Signed-off-by: Mark Brown --- sound/soc/codecs/max9877.c | 256 ++++++--------------------------------------- 1 file changed, 31 insertions(+), 225 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 7e2fe5023fb2..8505b401d3c4 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -30,189 +30,6 @@ static struct reg_default max9877_regs[] = { { 4, 0x49 }, }; -static int max9877_get_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int invert = mc->invert; - unsigned int val; - int ret; - - ret = regmap_read(regmap, reg, &val); - if (ret != 0) - return ret; - - ucontrol->value.integer.value[0] = (val >> shift) & mask; - - if (invert) - ucontrol->value.integer.value[0] = - mask - ucontrol->value.integer.value[0]; - - return 0; -} - -static int max9877_set_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int invert = mc->invert; - unsigned int val = (ucontrol->value.integer.value[0] & mask); - bool change; - int ret; - - if (invert) - val = mask - val; - - ret = regmap_update_bits_check(regmap, reg, mask << shift, - val << shift, &change); - if (ret != 0) - return ret; - - if (change) - return 1; - else - return 0; -} - -static int max9877_get_2reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int val; - int ret; - - ret = regmap_read(regmap, reg, &val); - if (ret != 0) - return ret; - ucontrol->value.integer.value[0] = (val >> shift) & mask; - - ret = regmap_read(regmap, reg2, &val); - if (ret != 0) - return ret; - ucontrol->value.integer.value[1] = (val >> shift) & mask; - - return 0; -} - -static int max9877_set_2reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int val = (ucontrol->value.integer.value[0] & mask); - unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - bool change1, change2; - int ret; - - ret = regmap_update_bits_check(regmap, reg, mask << shift, - val << shift, &change1); - if (ret != 0) - return ret; - - ret = regmap_update_bits_check(regmap, reg2, mask << shift, - val2 << shift, &change2); - if (ret != 0) - return ret; - - if (change1 || change2) - return 1; - else - return 0; -} - -static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int val; - int ret; - - ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); - if (ret != 0) - return ret; - - val &= MAX9877_OUTMODE_MASK; - if (val) - val--; - - ucontrol->value.integer.value[0] = val; - - return 0; -} - -static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int val; - bool change; - int ret; - - val = ucontrol->value.integer.value[0] + 1; - - ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, - MAX9877_OUTMODE_MASK, val, &change); - if (ret != 0) - return ret; - - if (change) - return 1; - else - return 0; -} - -static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int val; - int ret; - - ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); - if (ret != 0) - return ret; - - val &= MAX9877_OSC_MASK; - val >>= MAX9877_OSC_OFFSET; - - ucontrol->value.integer.value[0] = val; - - return 0; -} - -static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int val; - bool change; - int ret; - - val = ucontrol->value.integer.value[0] << MAX9877_OSC_OFFSET; - ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, - MAX9877_OSC_MASK, val, &change); - if (ret != 0) - return ret; - - if (change) - return 1; - else - return 0; -} - static const unsigned int max9877_pgain_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 1, TLV_DB_SCALE_ITEM(0, 900, 0), @@ -246,51 +63,40 @@ static const char *max9877_osc_mode[] = { }; static const struct soc_enum max9877_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode), - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), + SOC_ENUM_SINGLE(MAX9877_OUTPUT_MODE, 0, ARRAY_SIZE(max9877_out_mode), + max9877_out_mode), + SOC_ENUM_SINGLE(MAX9877_OUTPUT_MODE, MAX9877_OSC_OFFSET, + ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), }; static const struct snd_kcontrol_new max9877_controls[] = { - SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume", - MAX9877_INPUT_MODE, 0, 2, 0, - max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), - SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume", - MAX9877_INPUT_MODE, 2, 2, 0, - max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), - SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume", - MAX9877_SPK_VOLUME, 0, 31, 0, - max9877_get_reg, max9877_set_reg, max9877_output_tlv), - SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume", - MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, - max9877_get_2reg, max9877_set_2reg, max9877_output_tlv), - SOC_SINGLE_EXT("MAX9877 INB Stereo Switch", - MAX9877_INPUT_MODE, 4, 1, 1, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 INA Stereo Switch", - MAX9877_INPUT_MODE, 5, 1, 1, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch", - MAX9877_INPUT_MODE, 6, 1, 0, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch", - MAX9877_OUTPUT_MODE, 6, 1, 0, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch", - MAX9877_OUTPUT_MODE, 7, 1, 1, - max9877_get_reg, max9877_set_reg), - SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0], - max9877_get_out_mode, max9877_set_out_mode), - SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1], - max9877_get_osc_mode, max9877_set_osc_mode), + SOC_SINGLE_TLV("MAX9877 PGAINA Playback Volume", + MAX9877_INPUT_MODE, 0, 2, 0, max9877_pgain_tlv), + SOC_SINGLE_TLV("MAX9877 PGAINB Playback Volume", + MAX9877_INPUT_MODE, 2, 2, 0, max9877_pgain_tlv), + SOC_SINGLE_TLV("MAX9877 Amp Speaker Playback Volume", + MAX9877_SPK_VOLUME, 0, 31, 0, max9877_output_tlv), + SOC_DOUBLE_R_TLV("MAX9877 Amp HP Playback Volume", + MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, + max9877_output_tlv), + SOC_SINGLE("MAX9877 INB Stereo Switch", + MAX9877_INPUT_MODE, 4, 1, 1), + SOC_SINGLE("MAX9877 INA Stereo Switch", + MAX9877_INPUT_MODE, 5, 1, 1), + SOC_SINGLE("MAX9877 Zero-crossing detection Switch", + MAX9877_INPUT_MODE, 6, 1, 0), + SOC_SINGLE("MAX9877 Bypass Mode Switch", + MAX9877_OUTPUT_MODE, 6, 1, 0), + SOC_SINGLE("MAX9877 Shutdown Mode Switch", + MAX9877_OUTPUT_MODE, 7, 1, 1), + SOC_ENUM("MAX9877 Output Mode", max9877_enum[0]), + SOC_ENUM("MAX9877 Oscillator Mode", max9877_enum[1]), }; -/* This function is called from ASoC machine driver */ -int max9877_add_controls(struct snd_soc_codec *codec) -{ - return snd_soc_add_codec_controls(codec, max9877_controls, - ARRAY_SIZE(max9877_controls)); -} -EXPORT_SYMBOL_GPL(max9877_add_controls); +static const struct snd_soc_codec_driver max9877_codec = { + .controls = max9877_controls, + .num_controls = ARRAY_SIZE(max9877_controls), +}; static const struct regmap_config max9877_regmap = { .reg_bits = 8, @@ -314,12 +120,12 @@ static int max9877_i2c_probe(struct i2c_client *client, for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) regmap_write(regmap, max9877_regs[i].reg, max9877_regs[i].def); - return 0; + return snd_soc_register_codec(&client->dev, &max9877_codec, NULL, 0); } static int max9877_i2c_remove(struct i2c_client *client) { - regmap = NULL; + snd_soc_unregister_codec(&client->dev); return 0; } -- cgit v1.2.3-70-g09d2 From 5cf9da8aacbfaed72ada8c195859f49d5d7f5f6c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 13:32:03 +0100 Subject: ASoC: max9877: Add basic DAPM support This does not fully map the power control available within the device but it provides the hooks for routing signals through the device and allows automatic management of the device low power mode. Signed-off-by: Mark Brown --- sound/soc/codecs/max9877.c | 39 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 37 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 8505b401d3c4..29549cdbf4c1 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -87,15 +87,50 @@ static const struct snd_kcontrol_new max9877_controls[] = { MAX9877_INPUT_MODE, 6, 1, 0), SOC_SINGLE("MAX9877 Bypass Mode Switch", MAX9877_OUTPUT_MODE, 6, 1, 0), - SOC_SINGLE("MAX9877 Shutdown Mode Switch", - MAX9877_OUTPUT_MODE, 7, 1, 1), SOC_ENUM("MAX9877 Output Mode", max9877_enum[0]), SOC_ENUM("MAX9877 Oscillator Mode", max9877_enum[1]), }; +static const struct snd_soc_dapm_widget max9877_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("INA1"), +SND_SOC_DAPM_INPUT("INA2"), +SND_SOC_DAPM_INPUT("INB1"), +SND_SOC_DAPM_INPUT("INB2"), +SND_SOC_DAPM_INPUT("RXIN+"), +SND_SOC_DAPM_INPUT("RXIN-"), + +SND_SOC_DAPM_PGA("SHDN", MAX9877_OUTPUT_MODE, 7, 1, NULL, 0), + +SND_SOC_DAPM_OUTPUT("OUT+"), +SND_SOC_DAPM_OUTPUT("OUT-"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +}; + +static const struct snd_soc_dapm_route max9877_dapm_routes[] = { + { "SHDN", NULL, "INA1" }, + { "SHDN", NULL, "INA2" }, + { "SHDN", NULL, "INB1" }, + { "SHDN", NULL, "INB2" }, + + { "OUT+", NULL, "RXIN+" }, + { "OUT+", NULL, "SHDN" }, + + { "OUT-", NULL, "SHDN" }, + { "OUT-", NULL, "RXIN-" }, + + { "HPL", NULL, "SHDN" }, + { "HPR", NULL, "SHDN" }, +}; + static const struct snd_soc_codec_driver max9877_codec = { .controls = max9877_controls, .num_controls = ARRAY_SIZE(max9877_controls), + + .dapm_widgets = max9877_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9877_dapm_widgets), + .dapm_routes = max9877_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9877_dapm_routes), }; static const struct regmap_config max9877_regmap = { -- cgit v1.2.3-70-g09d2 From 7da493e9229c737c399886f57996f6bfd4454e21 Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Mon, 12 Aug 2013 15:19:51 +0530 Subject: ASoC: Samsung: I2S: Add quirks as driver data in I2S Samsung has different versions of I2S introduced in different platforms. Each version has some new support added for multichannel, secondary fifo, s/w reset control and internal mux for rclk src clk. Each newly added change has a quirk. So this patch adds all the required quirks as driver data and based on compatible string from dtsi fetches the quirks. Signed-off-by: Padmavathi Venna Reviewed-by: Tomasz Figa Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/samsung-i2s.txt | 18 +++---- sound/soc/samsung/i2s.c | 62 +++++++++++++--------- 2 files changed, 42 insertions(+), 38 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt index 025e66b85a43..25a0024d1b0a 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -2,7 +2,11 @@ Required SoC Specific Properties: -- compatible : "samsung,i2s-v5" +- compatible : should be one of the following. + - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with + secondary fifo, s/w reset control and internal mux for root clk src. + - reg: physical base address of the controller and length of memory mapped region. - dmas: list of DMA controller phandle and DMA request line ordered pairs. @@ -21,13 +25,6 @@ Required SoC Specific Properties: Optional SoC Specific Properties: -- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel - support, this flag is enabled. -- samsung,supports-rstclr: This flag should be set if I2S software reset bit - control is required. When this flag is set I2S software reset bit will be - enabled or disabled based on need. -- samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA, - then this flag is enabled. - samsung,idma-addr: Internal DMA register base address of the audio sub system(used in secondary sound source). - pinctrl-0: Should specify pin control groups used for this controller. @@ -36,7 +33,7 @@ Optional SoC Specific Properties: Example: i2s0: i2s@03830000 { - compatible = "samsung,i2s-v5"; + compatible = "samsung,s5pv210-i2s"; reg = <0x03830000 0x100>; dmas = <&pdma0 10 &pdma0 9 @@ -46,9 +43,6 @@ i2s0: i2s@03830000 { <&clock_audss EXYNOS_I2S_BUS>, <&clock_audss EXYNOS_SCLK_I2S>; clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - samsung,supports-6ch; - samsung,supports-rstclr; - samsung,supports-secdai; samsung,idma-addr = <0x03000000>; pinctrl-names = "default"; pinctrl-0 = <&i2s0_bus>; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 849ac0e225ca..3b4835a1bd23 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -40,6 +40,7 @@ enum samsung_dai_type { struct samsung_i2s_dai_data { int dai_type; + u32 quirks; }; struct i2s_dai { @@ -1032,18 +1033,18 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) static const struct of_device_id exynos_i2s_match[]; -static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) +static inline const struct samsung_i2s_dai_data *samsung_i2s_get_driver_data( + struct platform_device *pdev) { #ifdef CONFIG_OF - struct samsung_i2s_dai_data *data; if (pdev->dev.of_node) { const struct of_device_id *match; match = of_match_node(exynos_i2s_match, pdev->dev.of_node); - data = (struct samsung_i2s_dai_data *) match->data; - return data->dai_type; + return match->data; } else #endif - return platform_get_device_id(pdev)->driver_data; + return (struct samsung_i2s_dai_data *) + platform_get_device_id(pdev)->driver_data; } #ifdef CONFIG_PM_RUNTIME @@ -1074,13 +1075,13 @@ static int samsung_i2s_probe(struct platform_device *pdev) struct resource *res; u32 regs_base, quirks = 0, idma_addr = 0; struct device_node *np = pdev->dev.of_node; - enum samsung_dai_type samsung_dai_type; + const struct samsung_i2s_dai_data *i2s_dai_data; int ret = 0; /* Call during Seconday interface registration */ - samsung_dai_type = samsung_i2s_get_driver_data(pdev); + i2s_dai_data = samsung_i2s_get_driver_data(pdev); - if (samsung_dai_type == TYPE_SEC) { + if (i2s_dai_data->dai_type == TYPE_SEC) { sec_dai = dev_get_drvdata(&pdev->dev); if (!sec_dai) { dev_err(&pdev->dev, "Unable to get drvdata\n"); @@ -1129,15 +1130,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) idma_addr = i2s_cfg->idma_addr; } } else { - if (of_find_property(np, "samsung,supports-6ch", NULL)) - quirks |= QUIRK_PRI_6CHAN; - - if (of_find_property(np, "samsung,supports-secdai", NULL)) - quirks |= QUIRK_SEC_DAI; - - if (of_find_property(np, "samsung,supports-rstclr", NULL)) - quirks |= QUIRK_NEED_RSTCLR; - + quirks = i2s_dai_data->quirks; if (of_property_read_u32(np, "samsung,idma-addr", &idma_addr)) { if (quirks & QUIRK_SEC_DAI) { @@ -1250,27 +1243,44 @@ static int samsung_i2s_remove(struct platform_device *pdev) return 0; } +static const struct samsung_i2s_dai_data i2sv3_dai_type = { + .dai_type = TYPE_PRI, + .quirks = QUIRK_NO_MUXPSR, +}; + +static const struct samsung_i2s_dai_data i2sv5_dai_type = { + .dai_type = TYPE_PRI, + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR, +}; + +static const struct samsung_i2s_dai_data samsung_dai_type_pri = { + .dai_type = TYPE_PRI, +}; + +static const struct samsung_i2s_dai_data samsung_dai_type_sec = { + .dai_type = TYPE_SEC, +}; + static struct platform_device_id samsung_i2s_driver_ids[] = { { .name = "samsung-i2s", - .driver_data = TYPE_PRI, + .driver_data = (kernel_ulong_t)&samsung_dai_type_pri, }, { .name = "samsung-i2s-sec", - .driver_data = TYPE_SEC, + .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, }, {}, }; MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids); #ifdef CONFIG_OF -static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = { - [TYPE_PRI] = { TYPE_PRI }, - [TYPE_SEC] = { TYPE_SEC }, -}; - static const struct of_device_id exynos_i2s_match[] = { - { .compatible = "samsung,i2s-v5", - .data = &samsung_i2s_dai_data_array[TYPE_PRI], + { + .compatible = "samsung,s3c6410-i2s", + .data = &i2sv3_dai_type, + }, { + .compatible = "samsung,s5pv210-i2s", + .data = &i2sv5_dai_type, }, {}, }; -- cgit v1.2.3-70-g09d2 From 4ca0c0d4784fa82d68733f7793e3487023e12282 Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Mon, 12 Aug 2013 15:19:52 +0530 Subject: ASoC: Samsung: I2S: Modify the I2S driver to support I2S on Exynos5420 Exynos5420 added support for I2S TDM mode. For this, there are some register changes in the I2S controller. This patch adds the relevant register changes to support I2S in normal mode. This patch adds a quirk for TDM mode and if TDM mode is present all the relevent changes will be applied. Signed-off-by: Padmavathi Venna Reviewed-by: Tomasz Figa Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/samsung-i2s.txt | 4 ++ include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s-regs.h | 15 ++++ sound/soc/samsung/i2s.c | 81 +++++++++++++++++++--- 4 files changed, 93 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt index 25a0024d1b0a..7386d444ada1 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -6,6 +6,10 @@ Required SoC Specific Properties: - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with secondary fifo, s/w reset control and internal mux for root clk src. + - samsung,exynos5420-i2s: for 8/16/24bit multichannel(7.1) I2S with + secondary fifo, s/w reset control, internal mux for root clk src and + TDM support. TDM (Time division multiplexing) is to allow transfer of + multiple channel audio data on single data line. - reg: physical base address of the controller and length of memory mapped region. diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 88272591a895..9efc04dd255a 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -36,6 +36,7 @@ struct samsung_i2s { */ #define QUIRK_NO_MUXPSR (1 << 2) #define QUIRK_NEED_RSTCLR (1 << 3) +#define QUIRK_SUPPORTS_TDM (1 << 4) /* Quirks of the I2S controller */ u32 quirks; dma_addr_t idma_addr; diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h index 30513b7ede3a..821a50231002 100644 --- a/sound/soc/samsung/i2s-regs.h +++ b/sound/soc/samsung/i2s-regs.h @@ -31,6 +31,10 @@ #define I2SLVL1ADDR 0x34 #define I2SLVL2ADDR 0x38 #define I2SLVL3ADDR 0x3c +#define I2SSTR1 0x40 +#define I2SVER 0x44 +#define I2SFIC2 0x48 +#define I2STDM 0x4c #define CON_RSTCLR (1 << 31) #define CON_FRXOFSTATUS (1 << 26) @@ -117,6 +121,17 @@ #define MOD_BCLK_MASK 3 #define MOD_8BIT (1 << 0) +#define EXYNOS5420_MOD_LRP_SHIFT 15 +#define EXYNOS5420_MOD_SDF_SHIFT 6 +#define EXYNOS5420_MOD_RCLK_SHIFT 4 +#define EXYNOS5420_MOD_BCLK_SHIFT 0 +#define EXYNOS5420_MOD_BCLK_64FS 4 +#define EXYNOS5420_MOD_BCLK_96FS 5 +#define EXYNOS5420_MOD_BCLK_128FS 6 +#define EXYNOS5420_MOD_BCLK_192FS 7 +#define EXYNOS5420_MOD_BCLK_256FS 8 +#define EXYNOS5420_MOD_BCLK_MASK 0xf + #define MOD_CDCLKCON (1 << 12) #define PSR_PSREN (1 << 15) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 3b4835a1bd23..dd995a7ab55c 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -199,7 +199,12 @@ static inline bool is_manager(struct i2s_dai *i2s) /* Read RCLK of I2S (in multiples of LRCLK) */ static inline unsigned get_rfs(struct i2s_dai *i2s) { - u32 rfs = (readl(i2s->addr + I2SMOD) >> MOD_RCLK_SHIFT); + u32 rfs; + + if (i2s->quirks & QUIRK_SUPPORTS_TDM) + rfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_RCLK_SHIFT; + else + rfs = (readl(i2s->addr + I2SMOD) >> MOD_RCLK_SHIFT); rfs &= MOD_RCLK_MASK; switch (rfs) { @@ -214,8 +219,12 @@ static inline unsigned get_rfs(struct i2s_dai *i2s) static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) { u32 mod = readl(i2s->addr + I2SMOD); - int rfs_shift = MOD_RCLK_SHIFT; + int rfs_shift; + if (i2s->quirks & QUIRK_SUPPORTS_TDM) + rfs_shift = EXYNOS5420_MOD_RCLK_SHIFT; + else + rfs_shift = MOD_RCLK_SHIFT; mod &= ~(MOD_RCLK_MASK << rfs_shift); switch (rfs) { @@ -239,10 +248,22 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) /* Read Bit-Clock of I2S (in multiples of LRCLK) */ static inline unsigned get_bfs(struct i2s_dai *i2s) { - u32 bfs = readl(i2s->addr + I2SMOD) >> MOD_BCLK_SHIFT; - bfs &= MOD_BCLK_MASK; + u32 bfs; + + if (i2s->quirks & QUIRK_SUPPORTS_TDM) { + bfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_BCLK_SHIFT; + bfs &= EXYNOS5420_MOD_BCLK_MASK; + } else { + bfs = readl(i2s->addr + I2SMOD) >> MOD_BCLK_SHIFT; + bfs &= MOD_BCLK_MASK; + } switch (bfs) { + case 8: return 256; + case 7: return 192; + case 6: return 128; + case 5: return 96; + case 4: return 64; case 3: return 24; case 2: return 16; case 1: return 48; @@ -254,9 +275,22 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { u32 mod = readl(i2s->addr + I2SMOD); - int bfs_shift = MOD_BCLK_SHIFT; + int bfs_shift; + int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM; - mod &= ~(MOD_BCLK_MASK << bfs_shift); + if (i2s->quirks & QUIRK_SUPPORTS_TDM) { + bfs_shift = EXYNOS5420_MOD_BCLK_SHIFT; + mod &= ~(EXYNOS5420_MOD_BCLK_MASK << bfs_shift); + } else { + bfs_shift = MOD_BCLK_SHIFT; + mod &= ~(MOD_BCLK_MASK << bfs_shift); + } + + /* Non-TDM I2S controllers do not support BCLK > 48 * FS */ + if (!tdm && bfs > 48) { + dev_err(&i2s->pdev->dev, "Unsupported BCLK divider\n"); + return; + } switch (bfs) { case 48: @@ -271,6 +305,21 @@ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) case 16: mod |= (MOD_BCLK_16FS << bfs_shift); break; + case 64: + mod |= (EXYNOS5420_MOD_BCLK_64FS << bfs_shift); + break; + case 96: + mod |= (EXYNOS5420_MOD_BCLK_96FS << bfs_shift); + break; + case 128: + mod |= (EXYNOS5420_MOD_BCLK_128FS << bfs_shift); + break; + case 192: + mod |= (EXYNOS5420_MOD_BCLK_192FS << bfs_shift); + break; + case 256: + mod |= (EXYNOS5420_MOD_BCLK_256FS << bfs_shift); + break; default: dev_err(&i2s->pdev->dev, "Wrong BCLK Divider!\n"); return; @@ -496,10 +545,17 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, { struct i2s_dai *i2s = to_info(dai); u32 mod = readl(i2s->addr + I2SMOD); - int lrp_shift = MOD_LRP_SHIFT, sdf_shift = MOD_SDF_SHIFT; - int sdf_mask, lrp_rlow; + int lrp_shift, sdf_shift, sdf_mask, lrp_rlow; u32 tmp = 0; + if (i2s->quirks & QUIRK_SUPPORTS_TDM) { + lrp_shift = EXYNOS5420_MOD_LRP_SHIFT; + sdf_shift = EXYNOS5420_MOD_SDF_SHIFT; + } else { + lrp_shift = MOD_LRP_SHIFT; + sdf_shift = MOD_SDF_SHIFT; + } + sdf_mask = MOD_SDF_MASK << sdf_shift; lrp_rlow = MOD_LR_RLOW << lrp_shift; @@ -1253,6 +1309,12 @@ static const struct samsung_i2s_dai_data i2sv5_dai_type = { .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR, }; +static const struct samsung_i2s_dai_data i2sv6_dai_type = { + .dai_type = TYPE_PRI, + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | + QUIRK_SUPPORTS_TDM, +}; + static const struct samsung_i2s_dai_data samsung_dai_type_pri = { .dai_type = TYPE_PRI, }; @@ -1281,6 +1343,9 @@ static const struct of_device_id exynos_i2s_match[] = { }, { .compatible = "samsung,s5pv210-i2s", .data = &i2sv5_dai_type, + }, { + .compatible = "samsung,exynos5420-i2s", + .data = &i2sv6_dai_type, }, {}, }; -- cgit v1.2.3-70-g09d2 From 64efc5a0f272b370e5ae6e95ff3cd5023ce9fefc Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Wed, 14 Aug 2013 11:11:16 +0200 Subject: ASoC: samsung-ac97: simplify use of devm_ioremap_resource Remove unneeded error handling on the result of a call to platform_get_resource when the value is passed to devm_ioremap_resource. Move the call to platform_get_resource adjacent to the call to devm_ioremap_resource to make the connection between them more clear. A simplified version of the semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l; @@ - res = platform_get_resource(pdev, IORESOURCE_MEM, n); ... when != res - if (res == NULL) { ... \(goto l;\|return ret;\) } ... when != res + res = platform_get_resource(pdev, IORESOURCE_MEM, n); e = devm_ioremap_resource(e1, res); // Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/samsung/ac97.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 2dd623fa3882..c732df9a35b6 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -404,18 +404,13 @@ static int s3c_ac97_probe(struct platform_device *pdev) return -ENXIO; } - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem_res) { - dev_err(&pdev->dev, "Unable to get register resource\n"); - return -ENXIO; - } - irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); if (!irq_res) { dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); return -ENXIO; } + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); s3c_ac97.regs = devm_ioremap_resource(&pdev->dev, mem_res); if (IS_ERR(s3c_ac97.regs)) return PTR_ERR(s3c_ac97.regs); -- cgit v1.2.3-70-g09d2 From c324aac01be55253488aba9481523cd6f546f4ca Mon Sep 17 00:00:00 2001 From: Ma Haijun Date: Wed, 14 Aug 2013 09:15:38 +0800 Subject: ASoC: wm8960: Fix ADC volume bits Signed-off-by: Ma Haijun Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd1d2a7..368d39f19cdb 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -263,8 +263,8 @@ SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0), SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0), -SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH, - 0, 127, 0), +SOC_DOUBLE_R_TLV("ADC PCM Capture Volume", WM8960_LADC, WM8960_RADC, + 0, 255, 0, adc_tlv), SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume", WM8960_BYPASS1, 4, 7, 1, bypass_tlv), -- cgit v1.2.3-70-g09d2 From 4d8cfa4642f7d8fafa4d60f05dd34fe8c3b9fa45 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Mon, 12 Aug 2013 22:49:24 +0200 Subject: ASoC: mioa701_wm9713: Remove definition of ARRAY_AND_SIZE() Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/pxa/mioa701_wm9713.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 20fdce61060c..bbea7780eac6 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -56,8 +56,6 @@ #include "pxa2xx-ac97.h" #include "../codecs/wm9713.h" -#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x) - #define AC97_GPIO_PULL 0x58 /* Use GPIO8 for rear speaker amplifier */ -- cgit v1.2.3-70-g09d2 From c90c0d7a96e634a73ef1580f1d20993606545647 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 14 Aug 2013 14:24:16 -0600 Subject: ASoC: tegra: fix Tegra30 I2S capture parameter setup The Tegra30 I2S driver was writing the AHUB interface parameters to the playback path register rather than the capture path register. This caused the capture parameters not to be configured at all, so if capturing using non-HW-default parameters (e.g. 16-bit stereo rather than 8-bit mono) the audio would be corrupted. With this fixed, audio capture from an analog microphone works correctly on the Cardhu board. Cc: stable@vger.kernel.org Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index d04146cad61f..47565fd04505 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, reg = TEGRA30_I2S_CIF_RX_CTRL; } else { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - reg = TEGRA30_I2S_CIF_RX_CTRL; + reg = TEGRA30_I2S_CIF_TX_CTRL; } regmap_write(i2s->regmap, reg, val); -- cgit v1.2.3-70-g09d2 From 7ac0da8cd38cb09d0addf708a8abbb93cf325c68 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 14 Aug 2013 14:26:29 -0600 Subject: ASoC: tegra: support a Mic Jack in the Tegra+RT5640 machine driver Add a Mic Jack widget to the Tegra+RT5640 machine driver, and document this in the DT binding. This enables the DT to include the Mic Jack in the audio routing table, and hence enables capture of audio, in addition to the previously-working playback. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt | 1 + sound/soc/tegra/tegra_rt5640.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt index cba4f88bd9f0..dc6224994d69 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt @@ -16,6 +16,7 @@ Required properties: * Headphones * Speakers + * Mic Jack - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 08794f915a94..4511c5a875ec 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -99,6 +99,7 @@ static struct snd_soc_jack_gpio tegra_rt5640_hp_jack_gpio = { static const struct snd_soc_dapm_widget tegra_rt5640_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), }; static const struct snd_kcontrol_new tegra_rt5640_controls[] = { -- cgit v1.2.3-70-g09d2 From b4345006423d45622bc17198a598baefcea27c93 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Wed, 14 Aug 2013 11:11:19 +0200 Subject: ASoC: tegra20-ac97: simplify use of devm_ioremap_resource Remove unneeded error handling on the result of a call to platform_get_resource when the value is passed to devm_ioremap_resource. A simplified version of the semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l; @@ - res = platform_get_resource(pdev, IORESOURCE_MEM, n); ... when != res - if (res == NULL) { ... \(goto l;\|return ret;\) } ... when != res + res = platform_get_resource(pdev, IORESOURCE_MEM, n); e = devm_ioremap_resource(e1, res); // Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 87b845f4cd1b..964cedf454d1 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -334,12 +334,6 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - regs = devm_ioremap_resource(&pdev->dev, mem); if (IS_ERR(regs)) { ret = PTR_ERR(regs); -- cgit v1.2.3-70-g09d2 From b7ae6f31d8243ec684af16bc5c763eccdfabaec0 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 12 Aug 2013 10:42:37 +0200 Subject: ALSA: move dmaengine implementation from ASoC to ALSA core For the PXA DMA rework, we need the generic dmaengine implementation that currently lives in sound/soc for standalone (non-ASoC) AC'97 support. Move it to sound/core, and rename the Kconfig symbol. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/core/Kconfig | 3 + sound/core/Makefile | 3 + sound/core/pcm_dmaengine.c | 367 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/Kconfig | 5 +- sound/soc/Makefile | 4 - sound/soc/omap/Kconfig | 2 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/soc-dmaengine-pcm.c | 367 ------------------------------------------ sound/soc/spear/Kconfig | 2 +- 9 files changed, 377 insertions(+), 378 deletions(-) create mode 100644 sound/core/pcm_dmaengine.c delete mode 100644 sound/soc/soc-dmaengine-pcm.c (limited to 'sound') diff --git a/sound/core/Kconfig b/sound/core/Kconfig index c0c2f57a0d6f..94ce1c44ff83 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -6,6 +6,9 @@ config SND_PCM tristate select SND_TIMER +config SND_DMAENGINE_PCM + bool + config SND_HWDEP tristate diff --git a/sound/core/Makefile b/sound/core/Makefile index 43d4117428ac..5e890cfed423 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -13,6 +13,8 @@ snd-$(CONFIG_SND_JACK) += jack.o snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ pcm_memory.o +snd-pcm-dmaengine-objs := pcm_dmaengine.o + snd-page-alloc-y := memalloc.o snd-page-alloc-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o @@ -30,6 +32,7 @@ obj-$(CONFIG_SND_TIMER) += snd-timer.o obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o obj-$(CONFIG_SND_PCM) += snd-pcm.o snd-page-alloc.o +obj-$(CONFIG_SND_DMAENGINE_PCM) += snd-pcm-dmaengine.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o obj-$(CONFIG_SND_OSSEMUL) += oss/ diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c new file mode 100644 index 000000000000..aa924d9b7986 --- /dev/null +++ b/sound/core/pcm_dmaengine.c @@ -0,0 +1,367 @@ +/* + * Copyright (C) 2012, Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Based on: + * imx-pcm-dma-mx2.c, Copyright 2009 Sascha Hauer + * mxs-pcm.c, Copyright (C) 2011 Freescale Semiconductor, Inc. + * ep93xx-pcm.c, Copyright (C) 2006 Lennert Buytenhek + * Copyright (C) 2006 Applied Data Systems + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ +#include +#include +#include +#include +#include +#include +#include + +#include + +struct dmaengine_pcm_runtime_data { + struct dma_chan *dma_chan; + dma_cookie_t cookie; + + unsigned int pos; +}; + +static inline struct dmaengine_pcm_runtime_data *substream_to_prtd( + const struct snd_pcm_substream *substream) +{ + return substream->runtime->private_data; +} + +struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + + return prtd->dma_chan; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_get_chan); + +/** + * snd_hwparams_to_dma_slave_config - Convert hw_params to dma_slave_config + * @substream: PCM substream + * @params: hw_params + * @slave_config: DMA slave config + * + * This function can be used to initialize a dma_slave_config from a substream + * and hw_params in a dmaengine based PCM driver implementation. + */ +int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, + const struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config) +{ + enum dma_slave_buswidth buswidth; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; + break; + case SNDRV_PCM_FORMAT_S16_LE: + buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config->direction = DMA_MEM_TO_DEV; + slave_config->dst_addr_width = buswidth; + } else { + slave_config->direction = DMA_DEV_TO_MEM; + slave_config->src_addr_width = buswidth; + } + + slave_config->device_fc = false; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); + +/** + * snd_dmaengine_pcm_set_config_from_dai_data() - Initializes a dma slave config + * using DAI DMA data. + * @substream: PCM substream + * @dma_data: DAI DMA data + * @slave_config: DMA slave configuration + * + * Initializes the {dst,src}_addr, {dst,src}_maxburst, {dst,src}_addr_width and + * slave_id fields of the DMA slave config from the same fields of the DAI DMA + * data struct. The src and dst fields will be initialized depending on the + * direction of the substream. If the substream is a playback stream the dst + * fields will be initialized, if it is a capture stream the src fields will be + * initialized. The {dst,src}_addr_width field will only be initialized if the + * addr_width field of the DAI DMA data struct is not equal to + * DMA_SLAVE_BUSWIDTH_UNDEFINED. + */ +void snd_dmaengine_pcm_set_config_from_dai_data( + const struct snd_pcm_substream *substream, + const struct snd_dmaengine_dai_dma_data *dma_data, + struct dma_slave_config *slave_config) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config->dst_addr = dma_data->addr; + slave_config->dst_maxburst = dma_data->maxburst; + if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) + slave_config->dst_addr_width = dma_data->addr_width; + } else { + slave_config->src_addr = dma_data->addr; + slave_config->src_maxburst = dma_data->maxburst; + if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) + slave_config->src_addr_width = dma_data->addr_width; + } + + slave_config->slave_id = dma_data->slave_id; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); + +static void dmaengine_pcm_dma_complete(void *arg) +{ + struct snd_pcm_substream *substream = arg; + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + + prtd->pos += snd_pcm_lib_period_bytes(substream); + if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) + prtd->pos = 0; + + snd_pcm_period_elapsed(substream); +} + +static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dma_chan *chan = prtd->dma_chan; + struct dma_async_tx_descriptor *desc; + enum dma_transfer_direction direction; + unsigned long flags = DMA_CTRL_ACK; + + direction = snd_pcm_substream_to_dma_direction(substream); + + if (!substream->runtime->no_period_wakeup) + flags |= DMA_PREP_INTERRUPT; + + prtd->pos = 0; + desc = dmaengine_prep_dma_cyclic(chan, + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), direction, flags); + + if (!desc) + return -ENOMEM; + + desc->callback = dmaengine_pcm_dma_complete; + desc->callback_param = substream; + prtd->cookie = dmaengine_submit(desc); + + return 0; +} + +/** + * snd_dmaengine_pcm_trigger - dmaengine based PCM trigger implementation + * @substream: PCM substream + * @cmd: Trigger command + * + * Returns 0 on success, a negative error code otherwise. + * + * This function can be used as the PCM trigger callback for dmaengine based PCM + * driver implementations. + */ +int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = dmaengine_pcm_prepare_and_submit(substream); + if (ret) + return ret; + dma_async_issue_pending(prtd->dma_chan); + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dmaengine_resume(prtd->dma_chan); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dmaengine_pause(prtd->dma_chan); + break; + case SNDRV_PCM_TRIGGER_STOP: + dmaengine_terminate_all(prtd->dma_chan); + break; + default: + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); + +/** + * snd_dmaengine_pcm_pointer_no_residue - dmaengine based PCM pointer implementation + * @substream: PCM substream + * + * This function is deprecated and should not be used by new drivers, as its + * results may be unreliable. + */ +snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + return bytes_to_frames(substream->runtime, prtd->pos); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); + +/** + * snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation + * @substream: PCM substream + * + * This function can be used as the PCM pointer callback for dmaengine based PCM + * driver implementations. + */ +snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dma_tx_state state; + enum dma_status status; + unsigned int buf_size; + unsigned int pos = 0; + + status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); + if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { + buf_size = snd_pcm_lib_buffer_bytes(substream); + if (state.residue > 0 && state.residue <= buf_size) + pos = buf_size - state.residue; + } + + return bytes_to_frames(substream->runtime, pos); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); + +/** + * snd_dmaengine_pcm_request_channel - Request channel for the dmaengine PCM + * @filter_fn: Filter function used to request the DMA channel + * @filter_data: Data passed to the DMA filter function + * + * Returns NULL or the requested DMA channel. + * + * This function request a DMA channel for usage with dmaengine PCM. + */ +struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn, + void *filter_data) +{ + dma_cap_mask_t mask; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dma_cap_set(DMA_CYCLIC, mask); + + return dma_request_channel(mask, filter_fn, filter_data); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel); + +/** + * snd_dmaengine_pcm_open - Open a dmaengine based PCM substream + * @substream: PCM substream + * @chan: DMA channel to use for data transfers + * + * Returns 0 on success, a negative error code otherwise. + * + * The function should usually be called from the pcm open callback. Note that + * this function will use private_data field of the substream's runtime. So it + * is not availabe to your pcm driver implementation. + */ +int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, + struct dma_chan *chan) +{ + struct dmaengine_pcm_runtime_data *prtd; + int ret; + + if (!chan) + return -ENXIO; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + return -ENOMEM; + + prtd->dma_chan = chan; + + substream->runtime->private_data = prtd; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); + +/** + * snd_dmaengine_pcm_open_request_chan - Open a dmaengine based PCM substream and request channel + * @substream: PCM substream + * @filter_fn: Filter function used to request the DMA channel + * @filter_data: Data passed to the DMA filter function + * + * Returns 0 on success, a negative error code otherwise. + * + * This function will request a DMA channel using the passed filter function and + * data. The function should usually be called from the pcm open callback. Note + * that this function will use private_data field of the substream's runtime. So + * it is not availabe to your pcm driver implementation. + */ +int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, + dma_filter_fn filter_fn, void *filter_data) +{ + return snd_dmaengine_pcm_open(substream, + snd_dmaengine_pcm_request_channel(filter_fn, filter_data)); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan); + +/** + * snd_dmaengine_pcm_close - Close a dmaengine based PCM substream + * @substream: PCM substream + */ +int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + + kfree(prtd); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); + +/** + * snd_dmaengine_pcm_release_chan_close - Close a dmaengine based PCM substream and release channel + * @substream: PCM substream + * + * Releases the DMA channel associated with the PCM substream. + */ +int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + + dma_release_channel(prtd->dma_chan); + + return snd_dmaengine_pcm_close(substream); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close_release_chan); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 45eeaa9f7fec..5138b8493051 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -26,12 +26,9 @@ if SND_SOC config SND_SOC_AC97_BUS bool -config SND_SOC_DMAENGINE_PCM - bool - config SND_SOC_GENERIC_DMAENGINE_PCM bool - select SND_SOC_DMAENGINE_PCM + select SND_DMAENGINE_PCM # All the supported SoCs source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index bc0261476d7a..61a64d281905 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,10 +1,6 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o -ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),) -snd-soc-core-objs += soc-dmaengine-pcm.o -endif - ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o endif diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 9f5d55e6b17a..accd0ff0fbfc 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,7 +1,7 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" depends on ARCH_OMAP && DMA_OMAP - select SND_SOC_DMAENGINE_PCM + select SND_DMAENGINE_PCM config SND_OMAP_SOC_DMIC tristate diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index b35809467547..4db74a083db1 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -11,7 +11,7 @@ config SND_PXA2XX_SOC config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP - select SND_SOC_DMAENGINE_PCM + select SND_DMAENGINE_PCM select SND_ARM help Say Y if you want to add support for codecs attached to diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c deleted file mode 100644 index aa924d9b7986..000000000000 --- a/sound/soc/soc-dmaengine-pcm.c +++ /dev/null @@ -1,367 +0,0 @@ -/* - * Copyright (C) 2012, Analog Devices Inc. - * Author: Lars-Peter Clausen - * - * Based on: - * imx-pcm-dma-mx2.c, Copyright 2009 Sascha Hauer - * mxs-pcm.c, Copyright (C) 2011 Freescale Semiconductor, Inc. - * ep93xx-pcm.c, Copyright (C) 2006 Lennert Buytenhek - * Copyright (C) 2006 Applied Data Systems - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ -#include -#include -#include -#include -#include -#include -#include - -#include - -struct dmaengine_pcm_runtime_data { - struct dma_chan *dma_chan; - dma_cookie_t cookie; - - unsigned int pos; -}; - -static inline struct dmaengine_pcm_runtime_data *substream_to_prtd( - const struct snd_pcm_substream *substream) -{ - return substream->runtime->private_data; -} - -struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) -{ - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - - return prtd->dma_chan; -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_get_chan); - -/** - * snd_hwparams_to_dma_slave_config - Convert hw_params to dma_slave_config - * @substream: PCM substream - * @params: hw_params - * @slave_config: DMA slave config - * - * This function can be used to initialize a dma_slave_config from a substream - * and hw_params in a dmaengine based PCM driver implementation. - */ -int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, - const struct snd_pcm_hw_params *params, - struct dma_slave_config *slave_config) -{ - enum dma_slave_buswidth buswidth; - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; - break; - case SNDRV_PCM_FORMAT_S16_LE: - buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; - break; - case SNDRV_PCM_FORMAT_S18_3LE: - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: - buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - break; - default: - return -EINVAL; - } - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config->direction = DMA_MEM_TO_DEV; - slave_config->dst_addr_width = buswidth; - } else { - slave_config->direction = DMA_DEV_TO_MEM; - slave_config->src_addr_width = buswidth; - } - - slave_config->device_fc = false; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); - -/** - * snd_dmaengine_pcm_set_config_from_dai_data() - Initializes a dma slave config - * using DAI DMA data. - * @substream: PCM substream - * @dma_data: DAI DMA data - * @slave_config: DMA slave configuration - * - * Initializes the {dst,src}_addr, {dst,src}_maxburst, {dst,src}_addr_width and - * slave_id fields of the DMA slave config from the same fields of the DAI DMA - * data struct. The src and dst fields will be initialized depending on the - * direction of the substream. If the substream is a playback stream the dst - * fields will be initialized, if it is a capture stream the src fields will be - * initialized. The {dst,src}_addr_width field will only be initialized if the - * addr_width field of the DAI DMA data struct is not equal to - * DMA_SLAVE_BUSWIDTH_UNDEFINED. - */ -void snd_dmaengine_pcm_set_config_from_dai_data( - const struct snd_pcm_substream *substream, - const struct snd_dmaengine_dai_dma_data *dma_data, - struct dma_slave_config *slave_config) -{ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config->dst_addr = dma_data->addr; - slave_config->dst_maxburst = dma_data->maxburst; - if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) - slave_config->dst_addr_width = dma_data->addr_width; - } else { - slave_config->src_addr = dma_data->addr; - slave_config->src_maxburst = dma_data->maxburst; - if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) - slave_config->src_addr_width = dma_data->addr_width; - } - - slave_config->slave_id = dma_data->slave_id; -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); - -static void dmaengine_pcm_dma_complete(void *arg) -{ - struct snd_pcm_substream *substream = arg; - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - - prtd->pos += snd_pcm_lib_period_bytes(substream); - if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) - prtd->pos = 0; - - snd_pcm_period_elapsed(substream); -} - -static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) -{ - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - struct dma_chan *chan = prtd->dma_chan; - struct dma_async_tx_descriptor *desc; - enum dma_transfer_direction direction; - unsigned long flags = DMA_CTRL_ACK; - - direction = snd_pcm_substream_to_dma_direction(substream); - - if (!substream->runtime->no_period_wakeup) - flags |= DMA_PREP_INTERRUPT; - - prtd->pos = 0; - desc = dmaengine_prep_dma_cyclic(chan, - substream->runtime->dma_addr, - snd_pcm_lib_buffer_bytes(substream), - snd_pcm_lib_period_bytes(substream), direction, flags); - - if (!desc) - return -ENOMEM; - - desc->callback = dmaengine_pcm_dma_complete; - desc->callback_param = substream; - prtd->cookie = dmaengine_submit(desc); - - return 0; -} - -/** - * snd_dmaengine_pcm_trigger - dmaengine based PCM trigger implementation - * @substream: PCM substream - * @cmd: Trigger command - * - * Returns 0 on success, a negative error code otherwise. - * - * This function can be used as the PCM trigger callback for dmaengine based PCM - * driver implementations. - */ -int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - int ret; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - ret = dmaengine_pcm_prepare_and_submit(substream); - if (ret) - return ret; - dma_async_issue_pending(prtd->dma_chan); - break; - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - dmaengine_resume(prtd->dma_chan); - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dmaengine_pause(prtd->dma_chan); - break; - case SNDRV_PCM_TRIGGER_STOP: - dmaengine_terminate_all(prtd->dma_chan); - break; - default: - return -EINVAL; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); - -/** - * snd_dmaengine_pcm_pointer_no_residue - dmaengine based PCM pointer implementation - * @substream: PCM substream - * - * This function is deprecated and should not be used by new drivers, as its - * results may be unreliable. - */ -snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream) -{ - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - return bytes_to_frames(substream->runtime, prtd->pos); -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); - -/** - * snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation - * @substream: PCM substream - * - * This function can be used as the PCM pointer callback for dmaengine based PCM - * driver implementations. - */ -snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - struct dma_tx_state state; - enum dma_status status; - unsigned int buf_size; - unsigned int pos = 0; - - status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); - if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { - buf_size = snd_pcm_lib_buffer_bytes(substream); - if (state.residue > 0 && state.residue <= buf_size) - pos = buf_size - state.residue; - } - - return bytes_to_frames(substream->runtime, pos); -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); - -/** - * snd_dmaengine_pcm_request_channel - Request channel for the dmaengine PCM - * @filter_fn: Filter function used to request the DMA channel - * @filter_data: Data passed to the DMA filter function - * - * Returns NULL or the requested DMA channel. - * - * This function request a DMA channel for usage with dmaengine PCM. - */ -struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn, - void *filter_data) -{ - dma_cap_mask_t mask; - - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - dma_cap_set(DMA_CYCLIC, mask); - - return dma_request_channel(mask, filter_fn, filter_data); -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel); - -/** - * snd_dmaengine_pcm_open - Open a dmaengine based PCM substream - * @substream: PCM substream - * @chan: DMA channel to use for data transfers - * - * Returns 0 on success, a negative error code otherwise. - * - * The function should usually be called from the pcm open callback. Note that - * this function will use private_data field of the substream's runtime. So it - * is not availabe to your pcm driver implementation. - */ -int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, - struct dma_chan *chan) -{ - struct dmaengine_pcm_runtime_data *prtd; - int ret; - - if (!chan) - return -ENXIO; - - ret = snd_pcm_hw_constraint_integer(substream->runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - - prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); - if (!prtd) - return -ENOMEM; - - prtd->dma_chan = chan; - - substream->runtime->private_data = prtd; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); - -/** - * snd_dmaengine_pcm_open_request_chan - Open a dmaengine based PCM substream and request channel - * @substream: PCM substream - * @filter_fn: Filter function used to request the DMA channel - * @filter_data: Data passed to the DMA filter function - * - * Returns 0 on success, a negative error code otherwise. - * - * This function will request a DMA channel using the passed filter function and - * data. The function should usually be called from the pcm open callback. Note - * that this function will use private_data field of the substream's runtime. So - * it is not availabe to your pcm driver implementation. - */ -int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, - dma_filter_fn filter_fn, void *filter_data) -{ - return snd_dmaengine_pcm_open(substream, - snd_dmaengine_pcm_request_channel(filter_fn, filter_data)); -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan); - -/** - * snd_dmaengine_pcm_close - Close a dmaengine based PCM substream - * @substream: PCM substream - */ -int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) -{ - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - - kfree(prtd); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); - -/** - * snd_dmaengine_pcm_release_chan_close - Close a dmaengine based PCM substream and release channel - * @substream: PCM substream - * - * Releases the DMA channel associated with the PCM substream. - */ -int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream) -{ - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - - dma_release_channel(prtd->dma_chan); - - return snd_dmaengine_pcm_close(substream); -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close_release_chan); - -MODULE_LICENSE("GPL"); diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig index 3567d73b218e..0a53053495f3 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -1,6 +1,6 @@ config SND_SPEAR_SOC tristate - select SND_SOC_DMAENGINE_PCM + select SND_DMAENGINE_PCM config SND_SPEAR_SPDIF_OUT tristate -- cgit v1.2.3-70-g09d2 From 2023c90c3a2c4c1aeb7f47649367d551c676da07 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 12 Aug 2013 10:42:38 +0200 Subject: ASoC: pxa: pxa-ssp: add DT bindings The pxa ssp DAI acts as a user of a pxa ssp port, and needs an appropriate 'port' phandle in DT to reference the upstream. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 28 ++++++++++++++++ sound/soc/pxa/pxa-ssp.c | 37 ++++++++++++++++++---- 2 files changed, 59 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt new file mode 100644 index 000000000000..74c9ba6c2823 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt @@ -0,0 +1,28 @@ +Marvell PXA SSP CPU DAI bindings + +Required properties: + + compatible Must be "mrvl,pxa-ssp-dai" + port A phandle reference to a PXA ssp upstream device + +Example: + + /* upstream device */ + + ssp0: ssp@41000000 { + compatible = "mrvl,pxa3xx-ssp"; + reg = <0x41000000 0x40>; + interrupts = <24>; + clock-names = "pxa27x-ssp.0"; + dmas = <&dma 13 + &dma 14>; + dma-names = "rx", "tx"; + }; + + /* DAI as user */ + + ssp_dai0: ssp_dai@0 { + compatible = "mrvl,pxa-ssp-dai"; + port = <&ssp0>; + }; + diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6f4dd7543e82..19296f22cb28 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -21,6 +21,7 @@ #include #include #include +#include #include @@ -719,6 +720,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, static int pxa_ssp_probe(struct snd_soc_dai *dai) { + struct device *dev = dai->dev; struct ssp_priv *priv; int ret; @@ -726,10 +728,26 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) if (!priv) return -ENOMEM; - priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); - if (priv->ssp == NULL) { - ret = -ENODEV; - goto err_priv; + if (dev->of_node) { + struct device_node *ssp_handle; + + ssp_handle = of_parse_phandle(dev->of_node, "port", 0); + if (!ssp_handle) { + dev_err(dev, "unable to get 'port' phandle\n"); + return -ENODEV; + } + + priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio"); + if (priv->ssp == NULL) { + ret = -ENODEV; + goto err_priv; + } + } else { + priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); + if (priv->ssp == NULL) { + ret = -ENODEV; + goto err_priv; + } } priv->dai_fmt = (unsigned int) -1; @@ -798,6 +816,12 @@ static const struct snd_soc_component_driver pxa_ssp_component = { .name = "pxa-ssp", }; +#ifdef CONFIG_OF +static const struct of_device_id pxa_ssp_of_ids[] = { + { .compatible = "mrvl,pxa-ssp-dai" }, +}; +#endif + static int asoc_ssp_probe(struct platform_device *pdev) { return snd_soc_register_component(&pdev->dev, &pxa_ssp_component, @@ -812,8 +836,9 @@ static int asoc_ssp_remove(struct platform_device *pdev) static struct platform_driver asoc_ssp_driver = { .driver = { - .name = "pxa-ssp-dai", - .owner = THIS_MODULE, + .name = "pxa-ssp-dai", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pxa_ssp_of_ids), }, .probe = asoc_ssp_probe, -- cgit v1.2.3-70-g09d2 From d65a14587a9be853a887a1407db133df1fb68e29 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 12 Aug 2013 10:42:39 +0200 Subject: ASoC: pxa: use snd_dmaengine_dai_dma_data Use snd_dmaengine_dai_dma_data for passing the dma parameters from clients to the pxa pcm lib. This does no functional change, it's just an intermedia step to migrate the pxa bits over to dmaengine. The calculation of dcmd is a transition hack which will be removed again in a later patch. It's just there to make the transition more readable. Signed-off-by: Daniel Mack Acked-by: Mark Brown Signed-off-by: Mark Brown --- include/sound/pxa2xx-lib.h | 7 ----- sound/arm/pxa2xx-ac97.c | 26 ++++++++++-------- sound/arm/pxa2xx-pcm-lib.c | 52 +++++++++++++++++++++++++++++------ sound/arm/pxa2xx-pcm.c | 5 +++- sound/arm/pxa2xx-pcm.h | 6 ++-- sound/soc/pxa/mmp-pcm.c | 8 ++++-- sound/soc/pxa/mmp-sspa.c | 12 +++++--- sound/soc/pxa/pxa-ssp.c | 36 ++++++++---------------- sound/soc/pxa/pxa2xx-ac97.c | 67 +++++++++++++++++++++++---------------------- sound/soc/pxa/pxa2xx-i2s.c | 28 +++++++++---------- sound/soc/pxa/pxa2xx-pcm.c | 8 ++++-- 11 files changed, 142 insertions(+), 113 deletions(-) (limited to 'sound') diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 2fd3d251d9a5..56e818e4a1cb 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -6,13 +6,6 @@ /* PCM */ -struct pxa2xx_pcm_dma_params { - char *name; /* stream identifier */ - u32 dcmd; /* DMA descriptor dcmd field */ - volatile u32 *drcmr; /* the DMA request channel to use */ - u32 dev_addr; /* device physical address for DMA */ -}; - extern int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index ce431e6e07cf..5066a3768b28 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -14,12 +14,14 @@ #include #include #include +#include #include #include #include #include #include +#include #include #include @@ -41,20 +43,20 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_reset, }; -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = { - .name = "AC97 PCM out", - .dev_addr = __PREG(PCDR), - .drcmr = &DRCMR(12), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_ac97_pcm_out_req = 12; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { + .addr = __PREG(PCDR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_ac97_pcm_out_req, }; -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_in = { - .name = "AC97 PCM in", - .dev_addr = __PREG(PCDR), - .drcmr = &DRCMR(11), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_ac97_pcm_in_req = 11; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { + .addr = __PREG(PCDR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_ac97_pcm_in_req, }; static struct snd_pcm *pxa2xx_ac97_pcm; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 823359ed95e1..a61d7a9a995e 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -7,11 +7,13 @@ #include #include #include +#include #include #include #include #include +#include #include @@ -43,6 +45,35 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, size_t period = params_period_bytes(params); pxa_dma_desc *dma_desc; dma_addr_t dma_buff_phys, next_desc_phys; + u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG; + + /* temporary transition hack */ + switch (rtd->params->addr_width) { + case DMA_SLAVE_BUSWIDTH_1_BYTE: + dcmd |= DCMD_WIDTH1; + break; + case DMA_SLAVE_BUSWIDTH_2_BYTES: + dcmd |= DCMD_WIDTH2; + break; + case DMA_SLAVE_BUSWIDTH_4_BYTES: + dcmd |= DCMD_WIDTH4; + break; + default: + /* can't happen */ + break; + } + + switch (rtd->params->maxburst) { + case 8: + dcmd |= DCMD_BURST8; + break; + case 16: + dcmd |= DCMD_BURST16; + break; + case 32: + dcmd |= DCMD_BURST32; + break; + } snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = totsize; @@ -55,14 +86,14 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, dma_desc->ddadr = next_desc_phys; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dma_desc->dsadr = dma_buff_phys; - dma_desc->dtadr = rtd->params->dev_addr; + dma_desc->dtadr = rtd->params->addr; } else { - dma_desc->dsadr = rtd->params->dev_addr; + dma_desc->dsadr = rtd->params->addr; dma_desc->dtadr = dma_buff_phys; } if (period > totsize) period = totsize; - dma_desc->dcmd = rtd->params->dcmd | period | DCMD_ENDIRQEN; + dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN; dma_desc++; dma_buff_phys += period; } while (totsize -= period); @@ -76,8 +107,10 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - if (rtd && rtd->params && rtd->params->drcmr) - *rtd->params->drcmr = 0; + if (rtd && rtd->params && rtd->params->filter_data) { + unsigned long req = *(unsigned long *) rtd->params->filter_data; + DRCMR(req) = 0; + } snd_pcm_set_runtime_buffer(substream, NULL); return 0; @@ -136,6 +169,7 @@ EXPORT_SYMBOL(pxa2xx_pcm_pointer); int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; + unsigned long req; if (!prtd || !prtd->params) return 0; @@ -146,7 +180,8 @@ int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) DCSR(prtd->dma_ch) &= ~DCSR_RUN; DCSR(prtd->dma_ch) = 0; DCMD(prtd->dma_ch) = 0; - *prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD; + req = *(unsigned long *) prtd->params->filter_data; + DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD; return 0; } @@ -155,7 +190,6 @@ EXPORT_SYMBOL(__pxa2xx_pcm_prepare); void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) { struct snd_pcm_substream *substream = dev_id; - struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; int dcsr; dcsr = DCSR(dma_ch); @@ -164,8 +198,8 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) if (dcsr & DCSR_ENDINTR) { snd_pcm_period_elapsed(substream); } else { - printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", - rtd->params->name, dma_ch, dcsr); + printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n", + dma_ch, dcsr); snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); snd_pcm_stream_unlock(substream); diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index 26422a3584ea..69a2455b4472 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -11,8 +11,11 @@ */ #include +#include + #include #include +#include #include "pxa2xx-pcm.h" @@ -40,7 +43,7 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? client->playback_params : client->capture_params; - ret = pxa_request_dma(rtd->params->name, DMA_PRIO_LOW, + ret = pxa_request_dma("dma", DMA_PRIO_LOW, pxa2xx_pcm_dma_irq, substream); if (ret < 0) goto err2; diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h index 65f86b56ba42..2a8fc08d52a1 100644 --- a/sound/arm/pxa2xx-pcm.h +++ b/sound/arm/pxa2xx-pcm.h @@ -13,14 +13,14 @@ struct pxa2xx_runtime_data { int dma_ch; - struct pxa2xx_pcm_dma_params *params; + struct snd_dmaengine_dai_dma_data *params; pxa_dma_desc *dma_desc_array; dma_addr_t dma_desc_array_phys; }; struct pxa2xx_pcm_client { - struct pxa2xx_pcm_dma_params *playback_params; - struct pxa2xx_pcm_dma_params *capture_params; + struct snd_dmaengine_dai_dma_data *playback_params; + struct snd_dmaengine_dai_dma_data *capture_params; int (*startup)(struct snd_pcm_substream *); void (*shutdown)(struct snd_pcm_substream *); int (*prepare)(struct snd_pcm_substream *); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 5d57e071cdf5..9a97843ab09f 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -17,6 +17,8 @@ #include #include #include +#include + #include #include #include @@ -67,7 +69,7 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct pxa2xx_pcm_dma_params *dma_params; + struct snd_dmaengine_dai_dma_data *dma_params; struct dma_slave_config slave_config; int ret; @@ -80,10 +82,10 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, return ret; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = dma_params->dev_addr; + slave_config.dst_addr = dma_params->addr; slave_config.dst_maxburst = 4; } else { - slave_config.src_addr = dma_params->dev_addr; + slave_config.src_addr = dma_params->addr; slave_config.src_maxburst = 4; } diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 1605934d525e..41752a5fe3b0 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -27,12 +27,15 @@ #include #include #include +#include + #include #include #include #include #include #include +#include #include "mmp-sspa.h" /* @@ -40,7 +43,7 @@ */ struct sspa_priv { struct ssp_device *sspa; - struct pxa2xx_pcm_dma_params *dma_params; + struct snd_dmaengine_dai_dma_data *dma_params; struct clk *audio_clk; struct clk *sysclk; int dai_fmt; @@ -266,7 +269,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); struct ssp_device *sspa = sspa_priv->sspa; - struct pxa2xx_pcm_dma_params *dma_params; + struct snd_dmaengine_dai_dma_data *dma_params; u32 sspa_ctrl; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -309,7 +312,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, } dma_params = &sspa_priv->dma_params[substream->stream]; - dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + dma_params->addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? (sspa->phys_base + SSPA_TXD) : (sspa->phys_base + SSPA_RXD); snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params); @@ -425,7 +428,8 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) return -ENOMEM; priv->dma_params = devm_kzalloc(&pdev->dev, - 2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL); + 2 * sizeof(struct snd_dmaengine_dai_dma_data), + GFP_KERNEL); if (priv->dma_params == NULL) return -ENOMEM; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19296f22cb28..c0dcc3538e35 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -22,6 +22,7 @@ #include #include #include +#include #include @@ -31,9 +32,9 @@ #include #include #include +#include #include -#include #include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" @@ -80,27 +81,14 @@ static void pxa_ssp_disable(struct ssp_device *ssp) __raw_writel(sscr0, ssp->mmio_base + SSCR0); } -struct pxa2xx_pcm_dma_data { - struct pxa2xx_pcm_dma_params params; - char name[20]; -}; - static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4, - int out, struct pxa2xx_pcm_dma_params *dma_data) + int out, struct snd_dmaengine_dai_dma_data *dma) { - struct pxa2xx_pcm_dma_data *dma; - - dma = container_of(dma_data, struct pxa2xx_pcm_dma_data, params); - - snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, - width4 ? "32-bit" : "16-bit", out ? "out" : "in"); - - dma->params.name = dma->name; - dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx); - dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) : - (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | - (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; - dma->params.dev_addr = ssp->phys_base + SSDR; + dma->filter_data = out ? &ssp->drcmr_tx : &ssp->drcmr_rx; + dma->addr_width = width4 ? DMA_SLAVE_BUSWIDTH_4_BYTES : + DMA_SLAVE_BUSWIDTH_2_BYTES; + dma->maxburst = 16; + dma->addr = ssp->phys_base + SSDR; } static int pxa_ssp_startup(struct snd_pcm_substream *substream, @@ -108,7 +96,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, { struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; - struct pxa2xx_pcm_dma_data *dma; + struct snd_dmaengine_dai_dma_data *dma; int ret = 0; if (!cpu_dai->active) { @@ -116,10 +104,10 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } - dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; - snd_soc_dai_set_dma_data(cpu_dai, substream, &dma->params); + snd_soc_dai_set_dma_data(cpu_dai, substream, dma); return ret; } @@ -560,7 +548,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; - struct pxa2xx_pcm_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1475515712e6..f1059d999de6 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -14,15 +14,16 @@ #include #include #include +#include #include #include #include #include +#include #include #include -#include #include #include "pxa2xx-ac97.h" @@ -48,44 +49,44 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { - .name = "AC97 PCM Stereo out", - .dev_addr = __PREG(PCDR), - .drcmr = &DRCMR(12), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { + .addr = __PREG(PCDR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, }; -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = { - .name = "AC97 PCM Stereo in", - .dev_addr = __PREG(PCDR), - .drcmr = &DRCMR(11), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { + .addr = __PREG(PCDR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = { - .name = "AC97 Aux PCM (Slot 5) Mono out", - .dev_addr = __PREG(MODR), - .drcmr = &DRCMR(10), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, +static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { + .addr = __PREG(MODR), + .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .maxburst = 16, + .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = { - .name = "AC97 Aux PCM (Slot 5) Mono in", - .dev_addr = __PREG(MODR), - .drcmr = &DRCMR(9), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, +static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { + .addr = __PREG(MODR), + .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .maxburst = 16, + .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { - .name = "AC97 Mic PCM (Slot 6) Mono in", - .dev_addr = __PREG(MCDR), - .drcmr = &DRCMR(8), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, +static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8; +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { + .addr = __PREG(MCDR), + .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .maxburst = 16, + .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; #ifdef CONFIG_PM @@ -119,7 +120,7 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - struct pxa2xx_pcm_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dma_data = &pxa2xx_ac97_pcm_stereo_out; @@ -135,7 +136,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - struct pxa2xx_pcm_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dma_data = &pxa2xx_ac97_pcm_aux_mono_out; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index f7ca71664112..d5340a088858 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -23,9 +23,9 @@ #include #include #include +#include #include -#include #include #include "pxa2xx-i2s.h" @@ -82,20 +82,20 @@ static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; -static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { - .name = "I2S PCM Stereo out", - .dev_addr = __PREG(SADR), - .drcmr = &DRCMR(3), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; +static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { + .addr = __PREG(SADR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_i2s_pcm_stereo_out_req, }; -static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = { - .name = "I2S PCM Stereo in", - .dev_addr = __PREG(SADR), - .drcmr = &DRCMR(2), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST32 | DCMD_WIDTH4, +static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2; +static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { + .addr = __PREG(SADR), + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .maxburst = 32, + .filter_data = &pxa2xx_i2s_pcm_stereo_in_req, }; static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, @@ -163,7 +163,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct pxa2xx_pcm_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; BUG_ON(IS_ERR(clk_i2s)); clk_prepare_enable(clk_i2s); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index ecff116cb7b0..0aa2d695064a 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -12,10 +12,12 @@ #include #include +#include #include #include #include +#include #include "../../arm/pxa2xx-pcm.h" @@ -25,7 +27,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct pxa2xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct pxa2xx_pcm_dma_params *dma; + struct snd_dmaengine_dai_dma_data *dma; int ret; dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); @@ -39,7 +41,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, * with different params */ if (prtd->params == NULL) { prtd->params = dma; - ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW, + ret = pxa_request_dma("name", DMA_PRIO_LOW, pxa2xx_pcm_dma_irq, substream); if (ret < 0) return ret; @@ -47,7 +49,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, } else if (prtd->params != dma) { pxa_free_dma(prtd->dma_ch); prtd->params = dma; - ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW, + ret = pxa_request_dma("name", DMA_PRIO_LOW, pxa2xx_pcm_dma_irq, substream); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From a671468d336bc6c482ab04e88e6eaf38532270ee Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 12 Aug 2013 10:42:40 +0200 Subject: ASoC: pxa: pxa-ssp: set dma filter data from startup hook With the new dmaengine implementation, the filter_data parameter has to be set earlier, from pxa_ssp_startup(). Signed-off-by: Daniel Mack Acked-by: Mark Brown Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c0dcc3538e35..a3119a00d8fa 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -84,7 +84,6 @@ static void pxa_ssp_disable(struct ssp_device *ssp) static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4, int out, struct snd_dmaengine_dai_dma_data *dma) { - dma->filter_data = out ? &ssp->drcmr_tx : &ssp->drcmr_rx; dma->addr_width = width4 ? DMA_SLAVE_BUSWIDTH_4_BYTES : DMA_SLAVE_BUSWIDTH_2_BYTES; dma->maxburst = 16; @@ -107,6 +106,10 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; + + dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + &ssp->drcmr_tx : &ssp->drcmr_rx; + snd_soc_dai_set_dma_data(cpu_dai, substream, dma); return ret; -- cgit v1.2.3-70-g09d2 From c529ca4ab935c5a836bddec44cc80614df078a07 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 12 Aug 2013 10:42:41 +0200 Subject: ASoC: pxa: add DT bindings for pxa2xx-pcm The bindings do not carry any resources, as the module only registers the ASoC platform driver. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 +++++++++++++++ sound/soc/pxa/pxa2xx-pcm.c | 13 +++++++++++-- 2 files changed, 26 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt new file mode 100644 index 000000000000..551fbb8348c2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt @@ -0,0 +1,15 @@ +DT bindings for ARM PXA2xx PCM platform driver + +This is just a dummy driver that registers the PXA ASoC platform driver. +It does not have any resources assigned. + +Required properties: + + - compatible 'mrvl,pxa-pcm-audio' + +Example: + + pxa_pcm_audio: snd_soc_pxa_audio { + compatible = "mrvl,pxa-pcm-audio"; + }; + diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 0aa2d695064a..806da27b8b67 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include @@ -133,10 +134,18 @@ static int pxa2xx_soc_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id snd_soc_pxa_audio_match[] = { + { .compatible = "mrvl,pxa-pcm-audio" }, + { } +}; +#endif + static struct platform_driver pxa_pcm_driver = { .driver = { - .name = "pxa-pcm-audio", - .owner = THIS_MODULE, + .name = "pxa-pcm-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(snd_soc_pxa_audio_match), }, .probe = pxa2xx_soc_platform_probe, -- cgit v1.2.3-70-g09d2 From 5332e1d26fd182444e15e6481029347ab032d7cb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 18:29:44 +0100 Subject: ASoC: pcm1792a: Remove empty capture DAI stub These intialisations are just what will be done for static data anyway so remove them. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 72cf8353e812..c57d3a5665c5 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -162,10 +162,6 @@ static struct snd_soc_dai_driver pcm1792a_dai = { .channels_max = 2, .rates = PCM1792A_RATES, .formats = PCM1792A_FORMATS, }, - .capture = { - .channels_min = 0, - .channels_max = 0, - }, .ops = &pcm1792a_dai_ops, }; -- cgit v1.2.3-70-g09d2 From e7a5cb4223c86df522a97e21742aeef153db4ebb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 18:30:00 +0100 Subject: ASoC: pcm1792a: Add DAPM support Provide DAPM for the device, ensuring operation with DAPM required by the core and making it easier to hook up external hardware to it. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index c57d3a5665c5..2a8eccf64c76 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -154,6 +154,20 @@ static const struct snd_kcontrol_new pcm1792a_controls[] = { pcm1792a_dac_tlv), }; +static const struct snd_soc_dapm_widget pcm1792a_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("IOUTL+"), +SND_SOC_DAPM_OUTPUT("IOUTL-"), +SND_SOC_DAPM_OUTPUT("IOUTR+"), +SND_SOC_DAPM_OUTPUT("IOUTR-"), +}; + +static const struct snd_soc_dapm_route pcm1792a_dapm_routes[] = { + { "IOUTL+", NULL, "Playback" }, + { "IOUTL-", NULL, "Playback" }, + { "IOUTR+", NULL, "Playback" }, + { "IOUTR-", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver pcm1792a_dai = { .name = "pcm1792a-hifi", .playback = { @@ -184,6 +198,10 @@ static const struct regmap_config pcm1792a_regmap = { static struct snd_soc_codec_driver soc_codec_dev_pcm1792a = { .controls = pcm1792a_controls, .num_controls = ARRAY_SIZE(pcm1792a_controls), + .dapm_widgets = pcm1792a_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1792a_dapm_widgets), + .dapm_routes = pcm1792a_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1792a_dapm_routes), }; static int pcm1792a_spi_probe(struct spi_device *spi) -- cgit v1.2.3-70-g09d2 From b9281f99e30f795f28f6ea216289900b6e870d01 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 18:25:41 +0100 Subject: ASoC: pcm1681: Add DAPM support Provide DAPM for the device, ensuring operation with DAPM required by the core and making it easier to hook up external hardware to it. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1681.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 51b18662e6aa..651ce0923675 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -206,6 +206,28 @@ static const struct snd_soc_dai_ops pcm1681_dai_ops = { .digital_mute = pcm1681_digital_mute, }; +static const struct snd_soc_dapm_widget pcm1681_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUT1"), +SND_SOC_DAPM_OUTPUT("VOUT2"), +SND_SOC_DAPM_OUTPUT("VOUT3"), +SND_SOC_DAPM_OUTPUT("VOUT4"), +SND_SOC_DAPM_OUTPUT("VOUT5"), +SND_SOC_DAPM_OUTPUT("VOUT6"), +SND_SOC_DAPM_OUTPUT("VOUT7"), +SND_SOC_DAPM_OUTPUT("VOUT8"), +}; + +static const struct snd_soc_dapm_route pcm1681_dapm_routes[] = { + { "VOUT1", NULL, "Playback" }, + { "VOUT2", NULL, "Playback" }, + { "VOUT3", NULL, "Playback" }, + { "VOUT4", NULL, "Playback" }, + { "VOUT5", NULL, "Playback" }, + { "VOUT6", NULL, "Playback" }, + { "VOUT7", NULL, "Playback" }, + { "VOUT8", NULL, "Playback" }, +}; + static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); static const struct snd_kcontrol_new pcm1681_controls[] = { @@ -258,6 +280,10 @@ static const struct regmap_config pcm1681_regmap = { static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { .controls = pcm1681_controls, .num_controls = ARRAY_SIZE(pcm1681_controls), + .dapm_widgets = pcm1681_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1681_dapm_widgets), + .dapm_routes = pcm1681_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1681_dapm_routes), }; static const struct i2c_device_id pcm1681_i2c_id[] = { -- cgit v1.2.3-70-g09d2 From 903eb3187e1c322d2e6838fd0275f13a072c4b63 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 15 Aug 2013 16:03:52 +0200 Subject: ALSA: core: allow SND_DMAENGINE_PCM use from modules When users of SND_DMAENGINE_PCM are built as module, the config symbol SND_DMAENGINE_PCM must be tristate, otherwise the linker will fail. Signed-off-by: Daniel Mack Reported-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/core/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 94ce1c44ff83..313f22e9d929 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -7,7 +7,7 @@ config SND_PCM select SND_TIMER config SND_DMAENGINE_PCM - bool + tristate config SND_HWDEP tristate -- cgit v1.2.3-70-g09d2 From 1801928e0f99d94c55e33c584c5eb2ff5e246ee6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 Aug 2013 08:17:05 +0200 Subject: ALSA: hda - Add a fixup for Gateway LT27 Gateway LT27 needs a fixup for the inverted digital mic. Reported-by: "Nathanael D. Noblet" Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5b22bf958764..f303cd898515 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4339,6 +4339,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), -- cgit v1.2.3-70-g09d2 From 74b77b1510c90787d4e2e3a8412b85b235590ba5 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 16 Aug 2013 11:45:33 +0200 Subject: ASoC: imx-audmux: Move definitions to dt-bindings Move imx-audmux macro definitions to include/dt-bindings, so they can be used for devicetree. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- include/dt-bindings/sound/fsl-imx-audmux.h | 56 ++++++++++++++++++++++++++++++ sound/soc/fsl/imx-audmux.h | 52 +-------------------------- 2 files changed, 57 insertions(+), 51 deletions(-) create mode 100644 include/dt-bindings/sound/fsl-imx-audmux.h (limited to 'sound') diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h new file mode 100644 index 000000000000..50b09e96f247 --- /dev/null +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -0,0 +1,56 @@ +#ifndef __DT_FSL_IMX_AUDMUX_H +#define __DT_FSL_IMX_AUDMUX_H + +#define MX27_AUDMUX_HPCR1_SSI0 0 +#define MX27_AUDMUX_HPCR2_SSI1 1 +#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 +#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 +#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 +#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 + +#define MX31_AUDMUX_PORT1_SSI0 0 +#define MX31_AUDMUX_PORT2_SSI1 1 +#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 +#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 +#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 +#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 + +#define MX51_AUDMUX_PORT1_SSI0 0 +#define MX51_AUDMUX_PORT2_SSI1 1 +#define MX51_AUDMUX_PORT3 2 +#define MX51_AUDMUX_PORT4 3 +#define MX51_AUDMUX_PORT5 4 +#define MX51_AUDMUX_PORT6 5 +#define MX51_AUDMUX_PORT7 6 + +/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ +#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) +#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) +#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) +#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) +#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) +#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) +#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) +#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) +#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) +#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) + +/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ +#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) +#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) +#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) +#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) +#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) +#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) +#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) +#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) +#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) + +#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) +#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) +#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) + +#endif /* __DT_FSL_IMX_AUDMUX_H */ diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index b8ff44b9dafa..38a4209af7c6 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -1,57 +1,7 @@ #ifndef __IMX_AUDMUX_H #define __IMX_AUDMUX_H -#define MX27_AUDMUX_HPCR1_SSI0 0 -#define MX27_AUDMUX_HPCR2_SSI1 1 -#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 -#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 -#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 -#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 - -#define MX31_AUDMUX_PORT1_SSI0 0 -#define MX31_AUDMUX_PORT2_SSI1 1 -#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 -#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 -#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 -#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 -#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 - -#define MX51_AUDMUX_PORT1_SSI0 0 -#define MX51_AUDMUX_PORT2_SSI1 1 -#define MX51_AUDMUX_PORT3 2 -#define MX51_AUDMUX_PORT4 3 -#define MX51_AUDMUX_PORT5 4 -#define MX51_AUDMUX_PORT6 5 -#define MX51_AUDMUX_PORT7 6 - -/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) -#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) -#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) -#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) -#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) -#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) -#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) -#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) -#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) - -/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) -#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) -#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) -#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) -#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) -#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) -#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) -#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) -#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) - -#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) -#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) -#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) +#include int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr); -- cgit v1.2.3-70-g09d2 From 12201398fc9ad25f5c6568527e70c9a4bcf5fcee Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Aug 2013 12:13:50 +0100 Subject: ASoC: tlv320aic26: Remove direct use of internal I/O functions Use the core to do I/O rather than directly calling the driver operations in order to support further refactoring. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b192cd4705a0..a4f93608287e 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -174,9 +174,9 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG1, reg); reg = dval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg); /* Audio Control 3 (master mode, fsref rate) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3); @@ -185,13 +185,13 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, reg |= 0x0800; if (fsref == 48000) reg |= 0x2000; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Audio Control 1 (FSref divisor) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1); reg &= ~0x0fff; reg |= wlen | aic26->datfm | (divisor << 3) | divisor; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg); return 0; } @@ -212,7 +212,7 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) reg |= 0x8080; else reg &= ~0x8080; - aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg); + snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg); return 0; } @@ -348,7 +348,7 @@ static ssize_t aic26_keyclick_set(struct device *dev, val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); val |= 0x8000; - aic26_reg_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); return count; } @@ -368,20 +368,20 @@ static int aic26_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); /* Reset the codec to power on defaults */ - aic26_reg_write(codec, AIC26_REG_RESET, 0xBB00); + snd_soc_write(codec, AIC26_REG_RESET, 0xBB00); /* Power up CODEC */ - aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0); + snd_soc_write(codec, AIC26_REG_POWER_CTRL, 0); /* Audio Control 3 (master mode, fsref rate) */ - reg = aic26_reg_read(codec, AIC26_REG_AUDIO_CTRL3); + reg = snd_soc_read(codec, AIC26_REG_AUDIO_CTRL3); reg &= ~0xf800; reg |= 0x0800; /* set master mode */ - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Fill register cache */ for (i = 0; i < codec->driver->reg_cache_size; i++) - aic26_reg_read(codec, i); + snd_soc_read(codec, i); /* Register the sysfs files for debugging */ /* Create SysFS files */ -- cgit v1.2.3-70-g09d2 From c21bb9b1b7de87ee33c8ebf94a155be2aa551849 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Aug 2013 12:14:42 +0100 Subject: ASoC: tlv320aic26: Remove noisy print Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index a4f93608287e..93cf692a8db7 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -365,8 +365,6 @@ static int aic26_probe(struct snd_soc_codec *codec) aic26->codec = codec; - dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); - /* Reset the codec to power on defaults */ snd_soc_write(codec, AIC26_REG_RESET, 0xBB00); -- cgit v1.2.3-70-g09d2 From 4a11bc2fdd7f526c70e013366171d66f27656203 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Aug 2013 12:24:47 +0100 Subject: ASoC: tlv320aic26: Add basic DAPM support Provide external widgets for the CODEC to ensure the device continues to function with DAPM mandatory and to make it easier to hook the device up to other components. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 93cf692a8db7..7b8f3d965f43 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -119,6 +119,22 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } +static const struct snd_soc_dapm_widget tlv320aic26_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MICIN"), +SND_SOC_DAPM_INPUT("AUX"), + +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +}; + +static const struct snd_soc_dapm_route tlv320aic26_dapm_routes[] = { + { "Capture", NULL, "MICIN" }, + { "Capture", NULL, "AUX" }, + + { "HPL", NULL, "Playback" }, + { "HPR", NULL, "Playback" }, +}; + /* --------------------------------------------------------------------- * Digital Audio Interface Operations */ @@ -402,6 +418,10 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .write = aic26_reg_write, .reg_cache_size = AIC26_NUM_REGS, .reg_word_size = sizeof(u16), + .dapm_widgets = tlv320aic26_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets), + .dapm_routes = tlv320aic26_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes), }; /* --------------------------------------------------------------------- -- cgit v1.2.3-70-g09d2 From ac0b82b17894120b21937d0031fd0080b3ee2d83 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Aug 2013 11:54:51 +0100 Subject: ASoC: si476x: Add DAPM support This ensures the driver continues to work with DAPM mandatory and makes it easier to connect the device up to other components in the subsystem. Signed-off-by: Mark Brown Acked-by: Andrey Smirnov --- sound/soc/codecs/si476x.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 73e205c892a0..38f3b105c17d 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -102,6 +102,16 @@ static int si476x_codec_write(struct snd_soc_codec *codec, return err; } +static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +}; + +static const struct snd_soc_dapm_route si476x_dapm_routes[] = { + { "Capture", NULL, "LOUT" }, + { "Capture", NULL, "ROUT" }, +}; + static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -260,6 +270,10 @@ static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, .read = si476x_codec_read, .write = si476x_codec_write, + .dapm_widgets = si476x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), + .dapm_routes = si476x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(si476x_dapm_routes), }; static int si476x_platform_probe(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From 70a39b930f286d9d2b68391291dc02b85a2128e3 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sat, 17 Aug 2013 16:38:12 -0300 Subject: ASoC: fsl: Drop SND_SOC_FSL_UTILS from i.mx machine code SND_SOC_FSL_UTILS is only used by PowerPC machines, so let's drop it in the i.mx case. Signed-off-by: Fabio Estevam Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index e15f77197d0b..3a4808d376d0 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -175,7 +175,6 @@ config SND_SOC_IMX_WM8962 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS help Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. @@ -187,7 +186,6 @@ config SND_SOC_IMX_SGTL5000 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS help Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. -- cgit v1.2.3-70-g09d2 From 85fa532b6ef920b32598df86b194571a7059a77c Mon Sep 17 00:00:00 2001 From: Mike Dyer Date: Fri, 16 Aug 2013 18:36:28 +0100 Subject: ASoC: wm8960: Fix PLL register writes Bit 9 of PLL2,3 and 4 is reserved as '0'. The 24bit fractional part should be split across each register in 8bit chunks. Signed-off-by: Mike Dyer Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8960.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd1d2a7..5e5af898f7f8 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -857,9 +857,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, if (pll_div.k) { reg |= 0x20; - snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); - snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 16) & 0xff); + snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 8) & 0xff); + snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0xff); } snd_soc_write(codec, WM8960_PLL1, reg); -- cgit v1.2.3-70-g09d2 From ea67afc3fdbe9196d76ee79503a3809a54300b5a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 15 Aug 2013 11:53:28 +0100 Subject: ASoC: pcm3008: Use gpio_set_value_cansleep() We don't set the GPIO values from atomic context so support GPIOs that can't be controlled from atomic context. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index b883f99d6f9f..8b9b378bc0a7 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -54,8 +54,8 @@ static int pcm3008_soc_suspend(struct snd_soc_codec *codec) { struct pcm3008_setup_data *setup = codec->dev->platform_data; - gpio_set_value(setup->pdad_pin, 0); - gpio_set_value(setup->pdda_pin, 0); + gpio_set_value_cansleep(setup->pdad_pin, 0); + gpio_set_value_cansleep(setup->pdda_pin, 0); return 0; } @@ -64,8 +64,8 @@ static int pcm3008_soc_resume(struct snd_soc_codec *codec) { struct pcm3008_setup_data *setup = codec->dev->platform_data; - gpio_set_value(setup->pdad_pin, 1); - gpio_set_value(setup->pdda_pin, 1); + gpio_set_value_cansleep(setup->pdad_pin, 1); + gpio_set_value_cansleep(setup->pdda_pin, 1); return 0; } -- cgit v1.2.3-70-g09d2 From faaf36f21642a140715b7d6cf897ab4f4f5a924d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 15 Aug 2013 12:01:40 +0100 Subject: ASoC: pcm3008: Add DAPM support Make it possible to connect external devices to the CODEC and ensure continued operation with non-DAPM support removed from the core. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 8b9b378bc0a7..19f5028354f6 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,6 +28,22 @@ #include "pcm3008.h" +static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINL"), +SND_SOC_DAPM_INPUT("VINR"), + +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = { + { "PCM3008 Capture", NULL, "VINL" }, + { "PCM3008 Capture", NULL, "VINR" }, + + { "VOUTL", NULL, "PCM3008 Playback" }, + { "VOUTR", NULL, "PCM3008 Playback" }, +}; + #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -77,6 +93,10 @@ static int pcm3008_soc_resume(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { .suspend = pcm3008_soc_suspend, .resume = pcm3008_soc_resume, + .dapm_widgets = pcm3008_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets), + .dapm_routes = pcm3008_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm3008_dapm_routes), }; static int pcm3008_codec_probe(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From 4fc932c6d8c0d2715bb7f2a2f657230ea360af87 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 15 Aug 2013 12:04:28 +0100 Subject: ASoC: pcm3008: Manage DAC and ADC power with DAPM Rather than leaving the DAC and ADC active whenever the system is running manage their power with DAPM. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 72 +++++++++++++++++++++++++--------------------- 1 file changed, 39 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 19f5028354f6..b6618c4a7597 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,20 +28,53 @@ #include "pcm3008.h" +static int pcm3008_dac_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct pcm3008_setup_data *setup = codec->dev->platform_data; + + gpio_set_value_cansleep(setup->pdda_pin, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static int pcm3008_adc_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct pcm3008_setup_data *setup = codec->dev->platform_data; + + gpio_set_value_cansleep(setup->pdad_pin, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = { SND_SOC_DAPM_INPUT("VINL"), SND_SOC_DAPM_INPUT("VINR"), +SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, pcm3008_dac_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_ADC_E("ADC", NULL, SND_SOC_NOPM, 0, 0, pcm3008_adc_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("VOUTL"), SND_SOC_DAPM_OUTPUT("VOUTR"), }; static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = { - { "PCM3008 Capture", NULL, "VINL" }, - { "PCM3008 Capture", NULL, "VINR" }, + { "PCM3008 Capture", NULL, "ADC" }, + { "ADC", NULL, "VINL" }, + { "ADC", NULL, "VINR" }, - { "VOUTL", NULL, "PCM3008 Playback" }, - { "VOUTR", NULL, "PCM3008 Playback" }, + { "DAC", NULL, "PCM3008 Playback" }, + { "VOUTL", NULL, "DAC" }, + { "VOUTR", NULL, "DAC" }, }; #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -65,34 +98,7 @@ static struct snd_soc_dai_driver pcm3008_dai = { }, }; -#ifdef CONFIG_PM -static int pcm3008_soc_suspend(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - gpio_set_value_cansleep(setup->pdad_pin, 0); - gpio_set_value_cansleep(setup->pdda_pin, 0); - - return 0; -} - -static int pcm3008_soc_resume(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - gpio_set_value_cansleep(setup->pdad_pin, 1); - gpio_set_value_cansleep(setup->pdda_pin, 1); - - return 0; -} -#else -#define pcm3008_soc_suspend NULL -#define pcm3008_soc_resume NULL -#endif - static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { - .suspend = pcm3008_soc_suspend, - .resume = pcm3008_soc_resume, .dapm_widgets = pcm3008_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets), .dapm_routes = pcm3008_dapm_routes, @@ -128,13 +134,13 @@ static int pcm3008_codec_probe(struct platform_device *pdev) /* Configure PDAD GPIO. */ ret = devm_gpio_request_one(&pdev->dev, setup->pdad_pin, - GPIOF_OUT_INIT_HIGH, "codec_pdad"); + GPIOF_OUT_INIT_LOW, "codec_pdad"); if (ret != 0) return ret; /* Configure PDDA GPIO. */ ret = devm_gpio_request_one(&pdev->dev, setup->pdda_pin, - GPIOF_OUT_INIT_HIGH, "codec_pdda"); + GPIOF_OUT_INIT_LOW, "codec_pdda"); if (ret != 0) return ret; -- cgit v1.2.3-70-g09d2 From e29deb48189cea0be8b24f5912d8c21a18cb0244 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 18 Aug 2013 18:25:53 +0100 Subject: ASoC: wl1273: Add stub DAPM support In order to ensure that the device continues to work with DAPM support being mandatory provide stub DAPM widgets and routes. Note that the public information on the device appears to make no mention of the FM support the driver appears to have. Signed-off-by: Mark Brown --- sound/soc/codecs/wl1273.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 54cd3da09abd..b7ab2ef567c8 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -290,6 +290,18 @@ static const struct snd_kcontrol_new wl1273_controls[] = { snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put), }; +static const struct snd_soc_dapm_widget wl1273_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route wl1273_dapm_routes[] = { + { "Capture", NULL, "RX" }, + + { "TX", NULL, "Playback" }, +}; + static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -483,6 +495,11 @@ static int wl1273_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wl1273 = { .probe = wl1273_probe, .remove = wl1273_remove, + + .dapm_widgets = wl1273_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wl1273_dapm_widgets), + .dapm_routes = wl1273_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wl1273_dapm_routes), }; static int wl1273_platform_probe(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From 782fbaba36731d46820f3a4f358a7b46a9cd795c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 12:29:07 +0100 Subject: ASoC: cs4270: Add DAPM support This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8e4779812b96..83c835d9fd88 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -139,6 +139,22 @@ struct cs4270_private { struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; +static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), + +SND_SOC_DAPM_OUTPUT("AOUTL"), +SND_SOC_DAPM_OUTPUT("AOUTR"), +}; + +static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, + + { "AOUTA", NULL, "Playback" }, + { "AOUTB", NULL, "Playback" }, +}; + /** * struct cs4270_mode_ratios - clock ratio tables * @ratio: the ratio of MCLK to the sample rate @@ -612,6 +628,10 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .controls = cs4270_snd_controls, .num_controls = ARRAY_SIZE(cs4270_snd_controls), + .dapm_widgets = cs4270_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4270_dapm_widgets), + .dapm_routes = cs4270_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs4270_dapm_routes), }; /* -- cgit v1.2.3-70-g09d2 From 72a061f763c8af8ace650ccb1d01f484a6465608 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 18 Aug 2013 18:35:54 +0100 Subject: ASoC: wm8727: Add DAPM support In order to make the device easier to hook up to external components in system designs and ensure operation when DAPM support becomes mandatory add DAPM support. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8727.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 462f5e4d5c05..7b1a6d5c11c6 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -23,6 +23,16 @@ #include #include +static const struct snd_soc_dapm_widget wm8727_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route wm8727_dapm_routes[] = { + { "VOUTL", NULL, "Playback" }, + { "VOUTR", NULL, "Playback" }, +}; + /* * Note this is a simple chip with no configuration interface, sample rate is * determined automatically by examining the Master clock and Bit clock ratios @@ -43,7 +53,12 @@ static struct snd_soc_dai_driver wm8727_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8727; +static struct snd_soc_codec_driver soc_codec_dev_wm8727 = { + .dapm_widgets = wm8727_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8727_dapm_widgets), + .dapm_routes = wm8727_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8727_dapm_routes), +}; static int wm8727_probe(struct platform_device *pdev) { -- cgit v1.2.3-70-g09d2 From 226059e1cdbb5d747bd008eba114af0b1a4a621e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 18 Aug 2013 18:36:06 +0100 Subject: ASoC: wm8782: Add DAPM support In order to make the device easier to hook up to external components in system designs and ensure operation when DAPM support becomes mandatory add DAPM support. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8782.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index f1fdbf63abb4..8092495605ce 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -26,6 +26,16 @@ #include #include +static const struct snd_soc_dapm_widget wm8782_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route wm8782_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + static struct snd_soc_dai_driver wm8782_dai = { .name = "wm8782", .capture = { @@ -40,7 +50,12 @@ static struct snd_soc_dai_driver wm8782_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8782; +static struct snd_soc_codec_driver soc_codec_dev_wm8782 = { + .dapm_widgets = wm8782_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets), + .dapm_routes = wm8782_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8782_dapm_routes), +}; static int wm8782_probe(struct platform_device *pdev) { -- cgit v1.2.3-70-g09d2 From 3efd8a6f1a74b4bbf54c992e1cf23381c64de216 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Aug 2013 23:58:58 +0100 Subject: ASoC: wm5102: Add inputs for noise and mic mixers The noise and mic mixer inputs were not connected, do so. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f38c52d43b8d..8bbddc151aa8 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1423,9 +1423,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -1552,6 +1549,9 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), -- cgit v1.2.3-70-g09d2 From 66e7aa22c751af82567f9af82fe7e1254f751870 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Aug 2013 23:59:08 +0100 Subject: ASoC: wm5110: Add inputs for noise and mic mixers The noise and mic mixer inputs were not connected, do so. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 38e50c81a953..bbd64384ca1c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -841,9 +841,6 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -978,6 +975,9 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), -- cgit v1.2.3-70-g09d2 From c5efb38a1354890297aed2a7e197ec5b23ce966a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Aug 2013 23:59:19 +0100 Subject: ASoC: wm8997: Add inputs for noise and mic mixers The noise and mic mixer inputs were not connected, do so. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 0a43bac2f4e3..6ec3de3efa4f 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -774,9 +774,6 @@ static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -886,6 +883,9 @@ static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC2INT2"), -- cgit v1.2.3-70-g09d2 From d2a369cb53a3f3733800d5160d60f9a5271fe44c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:18:07 +0100 Subject: ASoC: ac97: Provide stub DAPM integration Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index ec7351803c24..8d9ba4ba4bfe 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -23,6 +23,16 @@ #include #include +static const struct snd_soc_dapm_widget ac97_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ac97_routes[] = { + { "AC97 Capture", NULL, "RX" }, + { "TX", NULL, "AC97 Playback" }, +}; + static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -117,6 +127,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, + + .dapm_widgets = ac97_widgets, + .num_dapm_widgets = ARRAY_SIZE(ac97_widgets), + .dapm_routes = ac97_routes, + .num_dapm_routes = ARRAY_SIZE(ac97_routes), }; static int ac97_probe(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From c34e51b12751c3e81c752b385f02a97bf3f862da Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:17:36 +0100 Subject: ASoC: hdmi: Provide stub DAPM integration Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index f0986b9f1939..68342b121c96 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -23,11 +23,20 @@ #define DRV_NAME "hdmi-audio-codec" -static struct snd_soc_codec_driver hdmi_codec; +static const struct snd_soc_dapm_widget hdmi_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route hdmi_routes[] = { + { "Capture", NULL, "RX" }, + { "TX", NULL, "Playback" }, +}; static struct snd_soc_dai_driver hdmi_codec_dai = { .name = "hdmi-hifi", .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | @@ -38,6 +47,7 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { SNDRV_PCM_FMTBIT_S24_LE, }, .capture = { + .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | @@ -50,6 +60,13 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { }; +static struct snd_soc_codec_driver hdmi_codec = { + .dapm_widgets = hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), + .dapm_routes = hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(hdmi_routes), +}; + static int hdmi_codec_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &hdmi_codec, -- cgit v1.2.3-70-g09d2 From b9dff9c3d2c6c4a9cdb936f263ed293274e2f05a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:13:14 +0100 Subject: ASoC: bt-sco: Add generic compatible string Provide a common compatible string for device trees to list as a fallback for simplicity. We don't currently have a binding document but let's not fix that right now... Signed-off-by: Mark Brown --- sound/soc/codecs/bt-sco.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index a081d9fcb166..5c040ce33f1d 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -50,6 +50,9 @@ static struct platform_device_id bt_sco_driver_ids[] = { { .name = "dfbmcs320", }, + { + .name = "bt-sco", + }, {}, }; MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); -- cgit v1.2.3-70-g09d2 From 5195ca4902fe0b2ae01eb43ce522b89163672804 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:16:19 +0100 Subject: ASoC: bt-sco: Provide stub DAPM integration Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown --- sound/soc/codecs/bt-sco.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index 5c040ce33f1d..c4cf0699e77f 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -15,15 +15,27 @@ #include +static const struct snd_soc_dapm_widget bt_sco_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route bt_sco_routes[] = { + { "Capture", NULL, "RX" }, + { "TX", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver bt_sco_dai = { .name = "bt-sco-pcm", .playback = { + .stream_name = "Playback", .channels_min = 1, .channels_max = 1, .rates = SNDRV_PCM_RATE_8000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { + .stream_name = "Capture", .channels_min = 1, .channels_max = 1, .rates = SNDRV_PCM_RATE_8000, @@ -31,7 +43,12 @@ static struct snd_soc_dai_driver bt_sco_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_bt_sco; +static struct snd_soc_codec_driver soc_codec_dev_bt_sco = { + .dapm_widgets = bt_sco_widgets, + .num_dapm_widgets = ARRAY_SIZE(bt_sco_widgets), + .dapm_routes = bt_sco_routes, + .num_dapm_routes = ARRAY_SIZE(bt_sco_routes), +}; static int bt_sco_probe(struct platform_device *pdev) { -- cgit v1.2.3-70-g09d2 From 2f6e3ba0e0645011cbbd0289e9082d8007141498 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:09:39 +0100 Subject: ASoC: spdif: Add stub DAPM widgets for Rx Ensure that the driver continues to work with mandatory DAPM. Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_receiver.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index 26d34744c677..e3501f40c7b3 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -23,13 +23,26 @@ #include #include +static const struct snd_soc_dapm_widget dir_widgets[] = { + SND_SOC_DAPM_INPUT("spdif-in"), +}; + +static const struct snd_soc_dapm_route dir_routes[] = { + { "Capture", NULL, "spdif-in" }, +}; + #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) -static struct snd_soc_codec_driver soc_codec_spdif_dir; +static struct snd_soc_codec_driver soc_codec_spdif_dir = { + .dapm_widgets = dir_widgets, + .num_dapm_widgets = ARRAY_SIZE(dir_widgets), + .dapm_routes = dir_routes, + .num_dapm_routes = ARRAY_SIZE(dir_routes), +}; static struct snd_soc_dai_driver dir_stub_dai = { .name = "dir-hifi", -- cgit v1.2.3-70-g09d2 From fc6061486534a8dfee02dd6b9dd523789abd9a3d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:10:08 +0100 Subject: ASoC: spdif: Remove duplicate const Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_transmitter.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index 4e96d10d61a2..a078aa31052a 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -33,7 +33,7 @@ static const struct snd_soc_dapm_widget dit_widgets[] = { SND_SOC_DAPM_OUTPUT("spdif-out"), }; -static const const struct snd_soc_dapm_route dit_routes[] = { +static const struct snd_soc_dapm_route dit_routes[] = { { "spdif-out", NULL, "Playback" }, }; -- cgit v1.2.3-70-g09d2 From 3c1c32d3765876b72570966c819fac4b8c646394 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Aug 2013 12:07:19 +0100 Subject: ASoC: imx: Add MODULE_LICENSE to DMA drivers Reported-by: Ben Hutchings Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-dma.c | 3 +++ sound/soc/fsl/imx-pcm-fiq.c | 2 ++ 2 files changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index f323ce09f881..4dc1296688e9 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include @@ -73,3 +74,5 @@ void imx_pcm_dma_exit(struct platform_device *pdev) snd_dmaengine_pcm_unregister(&pdev->dev); } EXPORT_SYMBOL_GPL(imx_pcm_dma_exit); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 3b2ba994beee..34043c55f2a6 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -408,3 +408,5 @@ void imx_pcm_fiq_exit(struct platform_device *pdev) snd_soc_unregister_platform(&pdev->dev); } EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit); + +MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From 37e6071787908fa9009cbd002c86402720becc5f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 20:33:20 +0100 Subject: ASoC: samsung: Check to see if we managed to allocate a channel Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/dma.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 21b79262010e..50c1eb669e90 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -176,6 +176,10 @@ static int dma_hw_params(struct snd_pcm_substream *substream, prtd->params->ch = prtd->params->ops->request( prtd->params->channel, &req, rtd->cpu_dai->dev, prtd->params->ch_name); + if (!prtd->params->ch) { + pr_err("Failed to allocate DMA channel\n"); + return -ENXIO; + } prtd->params->ops->config(prtd->params->ch, &config); } -- cgit v1.2.3-70-g09d2 From 85ff3c29d720fddddf35681bf8f244dfd91f66fa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 22:59:05 +0100 Subject: ASoC: samsung: Rename DMA platform registration functions The current naming with a simple asoc_ prefix is too generic for use in multiplatform kernels. Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/ac97.c | 4 ++-- sound/soc/samsung/dma.c | 8 ++++---- sound/soc/samsung/dma.h | 4 ++-- sound/soc/samsung/i2s.c | 6 +++--- 4 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index c732df9a35b6..2acf987844e8 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -457,7 +457,7 @@ static int s3c_ac97_probe(struct platform_device *pdev) if (ret) goto err5; - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret); goto err6; @@ -480,7 +480,7 @@ static int s3c_ac97_remove(struct platform_device *pdev) { struct resource *irq_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 50c1eb669e90..a0c67f60f594 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -437,17 +437,17 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; -int asoc_dma_platform_register(struct device *dev) +int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); } -EXPORT_SYMBOL_GPL(asoc_dma_platform_register); +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); -void asoc_dma_platform_unregister(struct device *dev) +void samsung_asoc_dma_platform_unregister(struct device *dev) { snd_soc_unregister_platform(dev); } -EXPORT_SYMBOL_GPL(asoc_dma_platform_unregister); +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_unregister); MODULE_AUTHOR("Ben Dooks, "); MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 189a7a6d5020..0e86315a3eaf 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,7 +22,7 @@ struct s3c_dma_params { char *ch_name; }; -int asoc_dma_platform_register(struct device *dev); -void asoc_dma_platform_unregister(struct device *dev); +int samsung_asoc_dma_platform_register(struct device *dev); +void samsung_asoc_dma_platform_unregister(struct device *dev); #endif diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index dd995a7ab55c..8200fc1b6d03 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1146,7 +1146,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) snd_soc_register_component(&sec_dai->pdev->dev, &samsung_i2s_component, &sec_dai->i2s_dai_drv, 1); - asoc_dma_platform_register(&pdev->dev); + samsung_asoc_dma_platform_register(&pdev->dev); return 0; } @@ -1263,7 +1263,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); - asoc_dma_platform_register(&pdev->dev); + samsung_asoc_dma_platform_register(&pdev->dev); return 0; err: @@ -1293,7 +1293,7 @@ static int samsung_i2s_remove(struct platform_device *pdev) i2s->pri_dai = NULL; i2s->sec_dai = NULL; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; -- cgit v1.2.3-70-g09d2 From 741a509f34d8d702f70d0ad99b8152c57d76961e Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Mon, 19 Aug 2013 17:05:55 +0200 Subject: ASoC: core: Generic ac97 link reset functions This patch adds generic ac97 reset functions using pincontrol and gpio parsed from devicetree. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/soc-ac97link.txt | 28 ++++ include/sound/soc.h | 2 + sound/soc/soc-core.c | 153 +++++++++++++++++++++ 3 files changed, 183 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/soc-ac97link.txt (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/soc-ac97link.txt b/Documentation/devicetree/bindings/sound/soc-ac97link.txt new file mode 100644 index 000000000000..80152a87f239 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/soc-ac97link.txt @@ -0,0 +1,28 @@ +AC97 link bindings + +These bindings can be included within any other device node. + +Required properties: + - pinctrl-names: Has to contain following states to setup the correct + pinmuxing for the used gpios: + "ac97-running": AC97-link is active + "ac97-reset": AC97-link reset state + "ac97-warm-reset": AC97-link warm reset state + - ac97-gpios: List of gpio phandles with args in the order ac97-sync, + ac97-sdata, ac97-reset + + +Example: + +ssi { + ... + + pinctrl-names = "default", "ac97-running", "ac97-reset", "ac97-warm-reset"; + pinctrl-0 = <&ac97link_running>; + pinctrl-1 = <&ac97link_running>; + pinctrl-2 = <&ac97link_reset>; + pinctrl-3 = <&ac97link_warm_reset>; + ac97-gpios = <&gpio3 20 0 &gpio3 22 0 &gpio3 28 0>; + + ... +}; diff --git a/include/sound/soc.h b/include/sound/soc.h index 6eabee7ec15a..c0ac3bc7b7f4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -468,6 +468,8 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); +int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, + struct platform_device *pdev); /* *Controls diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d82ee386eab5..b5c91f9aa160 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -30,9 +30,12 @@ #include #include #include +#include #include #include #include +#include +#include #include #include #include @@ -69,6 +72,16 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +struct snd_ac97_reset_cfg { + struct pinctrl *pctl; + struct pinctrl_state *pstate_reset; + struct pinctrl_state *pstate_warm_reset; + struct pinctrl_state *pstate_run; + int gpio_sdata; + int gpio_sync; + int gpio_reset; +}; + /* returns the minimum number of bytes needed to represent * a particular given value */ static int min_bytes_needed(unsigned long val) @@ -2080,6 +2093,117 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); +static struct snd_ac97_reset_cfg snd_ac97_rst_cfg; + +static void snd_soc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct pinctrl *pctl = snd_ac97_rst_cfg.pctl; + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_warm_reset); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 1); + + udelay(10); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0); + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run); + msleep(2); +} + +static void snd_soc_ac97_reset(struct snd_ac97 *ac97) +{ + struct pinctrl *pctl = snd_ac97_rst_cfg.pctl; + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_reset); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0); + gpio_direction_output(snd_ac97_rst_cfg.gpio_sdata, 0); + gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 0); + + udelay(10); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 1); + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run); + msleep(2); +} + +static int snd_soc_ac97_parse_pinctl(struct device *dev, + struct snd_ac97_reset_cfg *cfg) +{ + struct pinctrl *p; + struct pinctrl_state *state; + int gpio; + int ret; + + p = devm_pinctrl_get(dev); + if (IS_ERR(p)) { + dev_err(dev, "Failed to get pinctrl\n"); + return PTR_RET(p); + } + cfg->pctl = p; + + state = pinctrl_lookup_state(p, "ac97-reset"); + if (IS_ERR(state)) { + dev_err(dev, "Can't find pinctrl state ac97-reset\n"); + return PTR_RET(state); + } + cfg->pstate_reset = state; + + state = pinctrl_lookup_state(p, "ac97-warm-reset"); + if (IS_ERR(state)) { + dev_err(dev, "Can't find pinctrl state ac97-warm-reset\n"); + return PTR_RET(state); + } + cfg->pstate_warm_reset = state; + + state = pinctrl_lookup_state(p, "ac97-running"); + if (IS_ERR(state)) { + dev_err(dev, "Can't find pinctrl state ac97-running\n"); + return PTR_RET(state); + } + cfg->pstate_run = state; + + gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 0); + if (gpio < 0) { + dev_err(dev, "Can't find ac97-sync gpio\n"); + return gpio; + } + ret = devm_gpio_request(dev, gpio, "AC97 link sync"); + if (ret) { + dev_err(dev, "Failed requesting ac97-sync gpio\n"); + return ret; + } + cfg->gpio_sync = gpio; + + gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 1); + if (gpio < 0) { + dev_err(dev, "Can't find ac97-sdata gpio %d\n", gpio); + return gpio; + } + ret = devm_gpio_request(dev, gpio, "AC97 link sdata"); + if (ret) { + dev_err(dev, "Failed requesting ac97-sdata gpio\n"); + return ret; + } + cfg->gpio_sdata = gpio; + + gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); + if (gpio < 0) { + dev_err(dev, "Can't find ac97-reset gpio\n"); + return gpio; + } + ret = devm_gpio_request(dev, gpio, "AC97 link reset"); + if (ret) { + dev_err(dev, "Failed requesting ac97-reset gpio\n"); + return ret; + } + cfg->gpio_reset = gpio; + + return 0; +} + struct snd_ac97_bus_ops *soc_ac97_ops; EXPORT_SYMBOL_GPL(soc_ac97_ops); @@ -2097,6 +2221,35 @@ int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) } EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); +/** + * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * + * This function sets the reset and warm_reset properties of ops and parses + * the device node of pdev to get pinctrl states and gpio numbers to use. + */ +int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, + struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct snd_ac97_reset_cfg cfg; + int ret; + + ret = snd_soc_ac97_parse_pinctl(dev, &cfg); + if (ret) + return ret; + + ret = snd_soc_set_ac97_ops(ops); + if (ret) + return ret; + + ops->warm_reset = snd_soc_ac97_warm_reset; + ops->reset = snd_soc_ac97_reset; + + snd_ac97_rst_cfg = cfg; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset); + /** * snd_soc_free_ac97_codec - free AC97 codec device * @codec: audio codec -- cgit v1.2.3-70-g09d2 From 0783e648988a2ccef6eac9b1c376e7832e09cd94 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sat, 17 Aug 2013 18:13:00 -0300 Subject: ASoC: fsl: fsl_ssi: Fix the order of resources removal In fsl_ssi_remove() we need to remove the resources in the opposite order that they were acquired in probe. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0c072ff10875..3168998dcf1d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -928,14 +928,14 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx) imx_pcm_dma_exit(pdev); - clk_disable_unprepare(ssi_private->clk); - } snd_soc_unregister_component(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, &ssi_private->dev_attr); + if (ssi_private->ssi_on_imx) + clk_disable_unprepare(ssi_private->clk); irq_dispose_mapping(ssi_private->irq); - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- cgit v1.2.3-70-g09d2 From 673c24e957db4be85a12e4260ace12dea805fa97 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Mon, 19 Aug 2013 10:51:51 +0200 Subject: ASoC: omap: simplify platform_get_resource_byname/devm_ioremap_resource Remove unneeded error handling on the result of a call to platform_get_resource_byname when the value is passed to devm_ioremap_resource. In the case of omap-dmic.c, the error-handling code of devm_ioremap_resource is also corrected to include releasing the clock. A simplified version of the semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // @@ expression pdev,res,e,e1; expression ret != 0; identifier l; @@ res = platform_get_resource_byname(...); - if (res == NULL) { ... \(goto l;\|return ret;\) } e = devm_ioremap_resource(e1, res); // Signed-off-by: Julia Lawall Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 9 +++------ sound/soc/omap/omap-mcpdm.c | 3 --- 2 files changed, 3 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 4db1f8e6e172..12e566be3793 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -480,15 +480,12 @@ static int asoc_dmic_probe(struct platform_device *pdev) dmic->dma_data.filter_data = "up_link"; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); - if (!res) { - dev_err(dmic->dev, "invalid memory resource\n"); - ret = -ENODEV; + dmic->io_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(dmic->io_base)) { + ret = PTR_ERR(dmic->io_base); goto err_put_clk; } - dmic->io_base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(dmic->io_base)) - return PTR_ERR(dmic->io_base); ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component, &omap_dmic_dai, 1); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index a49dc52f8abc..90d2a7cd2563 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -480,9 +480,6 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dma_data[1].filter_data = "up_link"; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); - if (res == NULL) - return -ENOMEM; - mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(mcpdm->io_base)) return PTR_ERR(mcpdm->io_base); -- cgit v1.2.3-70-g09d2 From a1ce31388dfc954fa034e5e840f7323a81cb9e90 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 22 Aug 2013 13:30:15 +0530 Subject: ASoC: pxa: Remove duplicate inclusion of dmaengine.h dmaengine.h header file was included twice. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/pxa/mmp-pcm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 9a97843ab09f..8235e231d89c 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -17,7 +17,6 @@ #include #include #include -#include #include #include -- cgit v1.2.3-70-g09d2 From a2388a498ad2f85be01aca29e364abf427d9b53c Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 21 Aug 2013 11:13:16 +0800 Subject: ASoC: fsl: Add S/PDIF CPU DAI driver This patch implements a device-tree-only CPU DAI driver for Freescale S/PDIF controller that supports stereo playback and record feature. Signed-off-by: Nicolin Chen Acked-by: Stephen Warren Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,spdif.txt | 54 + sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_spdif.c | 1236 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 191 +++ 5 files changed, 1486 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt new file mode 100644 index 000000000000..f2ae335670f5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -0,0 +1,54 @@ +Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller + +The Freescale S/PDIF audio block is a stereo transceiver that allows the +processor to receive and transmit digital audio via an coaxial cable or +a fibre cable. + +Required properties: + + - compatible : Compatible list, must contain "fsl,imx35-spdif". + + - reg : Offset and length of the register set for the device. + + - interrupts : Contains the spdif interrupt. + + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + + - dma-names : Two dmas have to be defined, "tx" and "rx". + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "core" The core clock of spdif controller + "rxtx<0-7>" Clock source list for tx and rx clock. + This clock list should be identical to + the source list connecting to the spdif + clock mux in "SPDIF Transceiver Clock + Diagram" of SoC reference manual. It + can also be referred to TxClk_Source + bit of register SPDIF_STC. + +Example: + +spdif: spdif@02004000 { + compatible = "fsl,imx35-spdif"; + reg = <0x02004000 0x4000>; + interrupts = <0 52 0x04>; + dmas = <&sdma 14 18 0>, + <&sdma 15 18 0>; + dma-names = "rx", "tx"; + + clocks = <&clks 197>, <&clks 3>, + <&clks 197>, <&clks 107>, + <&clks 0>, <&clks 118>, + <&clks 62>, <&clks 139>, + <&clks 0>; + clock-names = "core", "rxtx0", + "rxtx1", "rxtx2", + "rxtx3", "rxtx4", + "rxtx5", "rxtx6", + "rxtx7"; + + status = "okay"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3a4808d376d0..cd088cc8c866 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,6 +1,9 @@ config SND_SOC_FSL_SSI tristate +config SND_SOC_FSL_SPDIF + tristate + config SND_SOC_FSL_UTILS tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d4b4aa8b5649..4b5970e014dd 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -12,9 +12,11 @@ obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale PowerPC SSI/DMA Platform Support snd-soc-fsl-ssi-objs := fsl_ssi.o +snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o +obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c new file mode 100644 index 000000000000..42a43820d993 --- /dev/null +++ b/sound/soc/fsl/fsl_spdif.c @@ -0,0 +1,1236 @@ +/* + * Freescale S/PDIF ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * Based on stmp3xxx_spdif_dai.c + * Vladimir Barinov + * Copyright 2008 SigmaTel, Inc + * Copyright 2008 Embedded Alley Solutions, Inc + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "fsl_spdif.h" +#include "imx-pcm.h" + +#define FSL_SPDIF_TXFIFO_WML 0x8 +#define FSL_SPDIF_RXFIFO_WML 0x8 + +#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC) +#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL | INT_URX_OV|\ + INT_QRX_FUL | INT_QRX_OV | INT_UQ_SYNC | INT_UQ_ERR |\ + INT_RXFIFO_RESYNC | INT_LOSS_LOCK | INT_DPLL_LOCKED) + +/* Index list for the values that has if (DPLL Locked) condition */ +static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb }; +#define SRPC_NODPLL_START1 0x5 +#define SRPC_NODPLL_START2 0xc + +#define DEFAULT_RXCLK_SRC 1 + +/* + * SPDIF control structure + * Defines channel status, subcode and Q sub + */ +struct spdif_mixer_control { + /* spinlock to access control data */ + spinlock_t ctl_lock; + + /* IEC958 channel tx status bit */ + unsigned char ch_status[4]; + + /* User bits */ + unsigned char subcode[2 * SPDIF_UBITS_SIZE]; + + /* Q subcode part of user bits */ + unsigned char qsub[2 * SPDIF_QSUB_SIZE]; + + /* Buffer offset for U/Q */ + u32 upos; + u32 qpos; + + /* Ready buffer index of the two buffers */ + u32 ready_buf; +}; + +struct fsl_spdif_priv { + struct spdif_mixer_control fsl_spdif_control; + struct snd_soc_dai_driver cpu_dai_drv; + struct platform_device *pdev; + struct regmap *regmap; + bool dpll_locked; + u8 txclk_div[SPDIF_TXRATE_MAX]; + u8 txclk_src[SPDIF_TXRATE_MAX]; + u8 rxclk_src; + struct clk *txclk[SPDIF_TXRATE_MAX]; + struct clk *rxclk; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct snd_dmaengine_dai_dma_data dma_params_rx; + + /* The name space will be allocated dynamically */ + char name[0]; +}; + + +/* DPLL locked and lock loss interrupt handler */ +static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv) +{ + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + u32 locked; + + regmap_read(regmap, REG_SPDIF_SRPC, &locked); + locked &= SRPC_DPLL_LOCKED; + + dev_dbg(&pdev->dev, "isr: Rx dpll %s \n", + locked ? "locked" : "loss lock"); + + spdif_priv->dpll_locked = locked ? true : false; +} + +/* Receiver found illegal symbol interrupt handler */ +static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv) +{ + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + + dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n"); + + if (!spdif_priv->dpll_locked) { + /* DPLL unlocked seems no audio stream */ + regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0); + } +} + +/* U/Q Channel receive register full */ +static void spdif_irq_uqrx_full(struct fsl_spdif_priv *spdif_priv, char name) +{ + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + u32 *pos, size, val, reg; + + switch (name) { + case 'U': + pos = &ctrl->upos; + size = SPDIF_UBITS_SIZE; + reg = REG_SPDIF_SRU; + break; + case 'Q': + pos = &ctrl->qpos; + size = SPDIF_QSUB_SIZE; + reg = REG_SPDIF_SRQ; + break; + default: + dev_err(&pdev->dev, "unsupported channel name\n"); + return; + } + + dev_dbg(&pdev->dev, "isr: %c Channel receive register full\n", name); + + if (*pos >= size * 2) { + *pos = 0; + } else if (unlikely((*pos % size) + 3 > size)) { + dev_err(&pdev->dev, "User bit receivce buffer overflow\n"); + return; + } + + regmap_read(regmap, reg, &val); + ctrl->subcode[*pos++] = val >> 16; + ctrl->subcode[*pos++] = val >> 8; + ctrl->subcode[*pos++] = val; +} + +/* U/Q Channel sync found */ +static void spdif_irq_uq_sync(struct fsl_spdif_priv *spdif_priv) +{ + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct platform_device *pdev = spdif_priv->pdev; + + dev_dbg(&pdev->dev, "isr: U/Q Channel sync found\n"); + + /* U/Q buffer reset */ + if (ctrl->qpos == 0) + return; + + /* Set ready to this buffer */ + ctrl->ready_buf = (ctrl->qpos - 1) / SPDIF_QSUB_SIZE + 1; +} + +/* U/Q Channel framing error */ +static void spdif_irq_uq_err(struct fsl_spdif_priv *spdif_priv) +{ + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + u32 val; + + dev_dbg(&pdev->dev, "isr: U/Q Channel framing error\n"); + + /* Read U/Q data to clear the irq and do buffer reset */ + regmap_read(regmap, REG_SPDIF_SRU, &val); + regmap_read(regmap, REG_SPDIF_SRQ, &val); + + /* Drop this U/Q buffer */ + ctrl->ready_buf = 0; + ctrl->upos = 0; + ctrl->qpos = 0; +} + +/* Get spdif interrupt status and clear the interrupt */ +static u32 spdif_intr_status_clear(struct fsl_spdif_priv *spdif_priv) +{ + struct regmap *regmap = spdif_priv->regmap; + u32 val, val2; + + regmap_read(regmap, REG_SPDIF_SIS, &val); + regmap_read(regmap, REG_SPDIF_SIE, &val2); + + regmap_write(regmap, REG_SPDIF_SIC, val & val2); + + return val; +} + +static irqreturn_t spdif_isr(int irq, void *devid) +{ + struct fsl_spdif_priv *spdif_priv = (struct fsl_spdif_priv *)devid; + struct platform_device *pdev = spdif_priv->pdev; + u32 sis; + + sis = spdif_intr_status_clear(spdif_priv); + + if (sis & INT_DPLL_LOCKED) + spdif_irq_dpll_lock(spdif_priv); + + if (sis & INT_TXFIFO_UNOV) + dev_dbg(&pdev->dev, "isr: Tx FIFO under/overrun\n"); + + if (sis & INT_TXFIFO_RESYNC) + dev_dbg(&pdev->dev, "isr: Tx FIFO resync\n"); + + if (sis & INT_CNEW) + dev_dbg(&pdev->dev, "isr: cstatus new\n"); + + if (sis & INT_VAL_NOGOOD) + dev_dbg(&pdev->dev, "isr: validity flag no good\n"); + + if (sis & INT_SYM_ERR) + spdif_irq_sym_error(spdif_priv); + + if (sis & INT_BIT_ERR) + dev_dbg(&pdev->dev, "isr: receiver found parity bit error\n"); + + if (sis & INT_URX_FUL) + spdif_irq_uqrx_full(spdif_priv, 'U'); + + if (sis & INT_URX_OV) + dev_dbg(&pdev->dev, "isr: U Channel receive register overrun\n"); + + if (sis & INT_QRX_FUL) + spdif_irq_uqrx_full(spdif_priv, 'Q'); + + if (sis & INT_QRX_OV) + dev_dbg(&pdev->dev, "isr: Q Channel receive register overrun\n"); + + if (sis & INT_UQ_SYNC) + spdif_irq_uq_sync(spdif_priv); + + if (sis & INT_UQ_ERR) + spdif_irq_uq_err(spdif_priv); + + if (sis & INT_RXFIFO_UNOV) + dev_dbg(&pdev->dev, "isr: Rx FIFO under/overrun\n"); + + if (sis & INT_RXFIFO_RESYNC) + dev_dbg(&pdev->dev, "isr: Rx FIFO resync\n"); + + if (sis & INT_LOSS_LOCK) + spdif_irq_dpll_lock(spdif_priv); + + /* FIXME: Write Tx FIFO to clear TxEm */ + if (sis & INT_TX_EM) + dev_dbg(&pdev->dev, "isr: Tx FIFO empty\n"); + + /* FIXME: Read Rx FIFO to clear RxFIFOFul */ + if (sis & INT_RXFIFO_FUL) + dev_dbg(&pdev->dev, "isr: Rx FIFO full\n"); + + return IRQ_HANDLED; +} + +static int spdif_softreset(struct fsl_spdif_priv *spdif_priv) +{ + struct regmap *regmap = spdif_priv->regmap; + u32 val, cycle = 1000; + + regmap_write(regmap, REG_SPDIF_SCR, SCR_SOFT_RESET); + + /* + * RESET bit would be cleared after finishing its reset procedure, + * which typically lasts 8 cycles. 1000 cycles will keep it safe. + */ + do { + regmap_read(regmap, REG_SPDIF_SCR, &val); + } while ((val & SCR_SOFT_RESET) && cycle--); + + if (cycle) + return 0; + else + return -EBUSY; +} + +static void spdif_set_cstatus(struct spdif_mixer_control *ctrl, + u8 mask, u8 cstatus) +{ + ctrl->ch_status[3] &= ~mask; + ctrl->ch_status[3] |= cstatus & mask; +} + +static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv) +{ + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + u32 ch_status; + + ch_status = (bitrev8(ctrl->ch_status[0]) << 16) | + (bitrev8(ctrl->ch_status[1]) << 8) | + bitrev8(ctrl->ch_status[2]); + regmap_write(regmap, REG_SPDIF_STCSCH, ch_status); + + dev_dbg(&pdev->dev, "STCSCH: 0x%06x\n", ch_status); + + ch_status = bitrev8(ctrl->ch_status[3]) << 16; + regmap_write(regmap, REG_SPDIF_STCSCL, ch_status); + + dev_dbg(&pdev->dev, "STCSCL: 0x%06x\n", ch_status); +} + +/* Set SPDIF PhaseConfig register for rx clock */ +static int spdif_set_rx_clksrc(struct fsl_spdif_priv *spdif_priv, + enum spdif_gainsel gainsel, int dpll_locked) +{ + struct regmap *regmap = spdif_priv->regmap; + u8 clksrc = spdif_priv->rxclk_src; + + if (clksrc >= SRPC_CLKSRC_MAX || gainsel >= GAINSEL_MULTI_MAX) + return -EINVAL; + + regmap_update_bits(regmap, REG_SPDIF_SRPC, + SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK, + SRPC_CLKSRC_SEL_SET(clksrc) | SRPC_GAINSEL_SET(gainsel)); + + return 0; +} + +static int spdif_set_sample_rate(struct snd_pcm_substream *substream, + int sample_rate) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + unsigned long csfs = 0; + u32 stc, mask, rate; + u8 clk, div; + int ret; + + switch (sample_rate) { + case 32000: + rate = SPDIF_TXRATE_32000; + csfs = IEC958_AES3_CON_FS_32000; + break; + case 44100: + rate = SPDIF_TXRATE_44100; + csfs = IEC958_AES3_CON_FS_44100; + break; + case 48000: + rate = SPDIF_TXRATE_48000; + csfs = IEC958_AES3_CON_FS_48000; + break; + default: + dev_err(&pdev->dev, "unsupported sample rate %d\n", sample_rate); + return -EINVAL; + } + + clk = spdif_priv->txclk_src[rate]; + if (clk >= STC_TXCLK_SRC_MAX) { + dev_err(&pdev->dev, "tx clock source is out of range\n"); + return -EINVAL; + } + + div = spdif_priv->txclk_div[rate]; + if (div == 0) { + dev_err(&pdev->dev, "the divisor can't be zero\n"); + return -EINVAL; + } + + /* + * The S/PDIF block needs a clock of 64 * fs * div. The S/PDIF block + * will divide by (div). So request 64 * fs * (div+1) which will + * get rounded. + */ + ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (div + 1)); + if (ret) { + dev_err(&pdev->dev, "failed to set tx clock rate\n"); + return ret; + } + + dev_dbg(&pdev->dev, "expected clock rate = %d\n", + (64 * sample_rate * div)); + dev_dbg(&pdev->dev, "actual clock rate = %ld\n", + clk_get_rate(spdif_priv->txclk[rate])); + + /* set fs field in consumer channel status */ + spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs); + + /* select clock source and divisor */ + stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | STC_TXCLK_DIV(div); + mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | STC_TXCLK_DIV_MASK; + regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc); + + dev_dbg(&pdev->dev, "set sample rate to %d\n", sample_rate); + + return 0; +} + +int fsl_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct platform_device *pdev = spdif_priv->pdev; + struct regmap *regmap = spdif_priv->regmap; + u32 scr, mask, i; + int ret; + + /* Reset module and interrupts only for first initialization */ + if (!cpu_dai->active) { + ret = spdif_softreset(spdif_priv); + if (ret) { + dev_err(&pdev->dev, "failed to soft reset\n"); + return ret; + } + + /* Disable all the interrupts */ + regmap_update_bits(regmap, REG_SPDIF_SIE, 0xffffff, 0); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + scr = SCR_TXFIFO_AUTOSYNC | SCR_TXFIFO_CTRL_NORMAL | + SCR_TXSEL_NORMAL | SCR_USRC_SEL_CHIP | + SCR_TXFIFO_FSEL_IF8; + mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | + SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | + SCR_TXFIFO_FSEL_MASK; + for (i = 0; i < SPDIF_TXRATE_MAX; i++) + clk_prepare_enable(spdif_priv->txclk[i]); + } else { + scr = SCR_RXFIFO_FSEL_IF8 | SCR_RXFIFO_AUTOSYNC; + mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK| + SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK; + clk_prepare_enable(spdif_priv->rxclk); + } + regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); + + /* Power up SPDIF module */ + regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, 0); + + return 0; +} + +static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 scr, mask, i; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + scr = 0; + mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | + SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | + SCR_TXFIFO_FSEL_MASK; + for (i = 0; i < SPDIF_TXRATE_MAX; i++) + clk_disable_unprepare(spdif_priv->txclk[i]); + } else { + scr = SCR_RXFIFO_OFF | SCR_RXFIFO_CTL_ZERO; + mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK| + SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK; + clk_disable_unprepare(spdif_priv->rxclk); + } + regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); + + /* Power down SPDIF module only if tx&rx are both inactive */ + if (!cpu_dai->active) { + spdif_intr_status_clear(spdif_priv); + regmap_update_bits(regmap, REG_SPDIF_SCR, + SCR_LOW_POWER, SCR_LOW_POWER); + } +} + +static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + struct platform_device *pdev = spdif_priv->pdev; + u32 sample_rate = params_rate(params); + int ret = 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = spdif_set_sample_rate(substream, sample_rate); + if (ret) { + dev_err(&pdev->dev, "%s: set sample rate failed: %d\n", + __func__, sample_rate); + return ret; + } + spdif_set_cstatus(ctrl, IEC958_AES3_CON_CLOCK, + IEC958_AES3_CON_CLOCK_1000PPM); + spdif_write_channel_status(spdif_priv); + } else { + /* Setup rx clock source */ + ret = spdif_set_rx_clksrc(spdif_priv, SPDIF_DEFAULT_GAINSEL, 1); + } + + return ret; +} + +static int fsl_spdif_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + int is_playack = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + u32 intr = is_playack ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE; + u32 dmaen = is_playack ? SCR_DMA_TX_EN : SCR_DMA_RX_EN;; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + regmap_update_bits(regmap, REG_SPDIF_SIE, intr, intr); + regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, dmaen); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, 0); + regmap_update_bits(regmap, REG_SPDIF_SIE, intr, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +struct snd_soc_dai_ops fsl_spdif_dai_ops = { + .startup = fsl_spdif_startup, + .hw_params = fsl_spdif_hw_params, + .trigger = fsl_spdif_trigger, + .shutdown = fsl_spdif_shutdown, +}; + + +/* + * ============================================ + * FSL SPDIF IEC958 controller(mixer) functions + * + * Channel status get/put control + * User bit value get/put control + * Valid bit value get control + * DPLL lock status get control + * User bit sync mode selection control + * ============================================ + */ + +static int fsl_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int fsl_spdif_pb_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uvalue) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + + uvalue->value.iec958.status[0] = ctrl->ch_status[0]; + uvalue->value.iec958.status[1] = ctrl->ch_status[1]; + uvalue->value.iec958.status[2] = ctrl->ch_status[2]; + uvalue->value.iec958.status[3] = ctrl->ch_status[3]; + + return 0; +} + +static int fsl_spdif_pb_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uvalue) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + + ctrl->ch_status[0] = uvalue->value.iec958.status[0]; + ctrl->ch_status[1] = uvalue->value.iec958.status[1]; + ctrl->ch_status[2] = uvalue->value.iec958.status[2]; + ctrl->ch_status[3] = uvalue->value.iec958.status[3]; + + spdif_write_channel_status(spdif_priv); + + return 0; +} + +/* Get channel status from SPDIF_RX_CCHAN register */ +static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 cstatus, val; + + regmap_read(regmap, REG_SPDIF_SIS, &val); + if (!(val & INT_CNEW)) { + return -EAGAIN; + } + + regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus); + ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF; + ucontrol->value.iec958.status[1] = (cstatus >> 8) & 0xFF; + ucontrol->value.iec958.status[2] = cstatus & 0xFF; + + regmap_read(regmap, REG_SPDIF_SRCSL, &cstatus); + ucontrol->value.iec958.status[3] = (cstatus >> 16) & 0xFF; + ucontrol->value.iec958.status[4] = (cstatus >> 8) & 0xFF; + ucontrol->value.iec958.status[5] = cstatus & 0xFF; + + /* Clear intr */ + regmap_write(regmap, REG_SPDIF_SIC, INT_CNEW); + + return 0; +} + +/* + * Get User bits (subcode) from chip value which readed out + * in UChannel register. + */ +static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + unsigned long flags; + int ret = 0; + + spin_lock_irqsave(&ctrl->ctl_lock, flags); + if (ctrl->ready_buf) { + int idx = (ctrl->ready_buf - 1) * SPDIF_UBITS_SIZE; + memcpy(&ucontrol->value.iec958.subcode[0], + &ctrl->subcode[idx], SPDIF_UBITS_SIZE); + } else { + ret = -EAGAIN; + } + spin_unlock_irqrestore(&ctrl->ctl_lock, flags); + + return ret; +} + +/* Q-subcode infomation. The byte size is SPDIF_UBITS_SIZE/8 */ +static int fsl_spdif_qinfo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = SPDIF_QSUB_SIZE; + + return 0; +} + +/* Get Q subcode from chip value which readed out in QChannel register */ +static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; + unsigned long flags; + int ret = 0; + + spin_lock_irqsave(&ctrl->ctl_lock, flags); + if (ctrl->ready_buf) { + int idx = (ctrl->ready_buf - 1) * SPDIF_QSUB_SIZE; + memcpy(&ucontrol->value.bytes.data[0], + &ctrl->qsub[idx], SPDIF_QSUB_SIZE); + } else { + ret = -EAGAIN; + } + spin_unlock_irqrestore(&ctrl->ctl_lock, flags); + + return ret; +} + +/* Valid bit infomation */ +static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +/* Get valid good bit from interrupt status register */ +static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 val; + + val = regmap_read(regmap, REG_SPDIF_SIS, &val); + ucontrol->value.integer.value[0] = (val & INT_VAL_NOGOOD) != 0; + regmap_write(regmap, REG_SPDIF_SIC, INT_VAL_NOGOOD); + + return 0; +} + +/* DPLL lock infomation */ +static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 16000; + uinfo->value.integer.max = 96000; + + return 0; +} + +static u32 gainsel_multi[GAINSEL_MULTI_MAX] = { + 24, 16, 12, 8, 6, 4, 3, +}; + +/* Get RX data clock rate given the SPDIF bus_clk */ +static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv, + enum spdif_gainsel gainsel) +{ + struct regmap *regmap = spdif_priv->regmap; + struct platform_device *pdev = spdif_priv->pdev; + u64 tmpval64, busclk_freq = 0; + u32 freqmeas, phaseconf; + u8 clksrc; + + regmap_read(regmap, REG_SPDIF_SRFM, &freqmeas); + regmap_read(regmap, REG_SPDIF_SRPC, &phaseconf); + + clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf; + if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) { + /* Get bus clock from system */ + busclk_freq = clk_get_rate(spdif_priv->rxclk); + } + + /* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */ + tmpval64 = (u64) busclk_freq * freqmeas; + do_div(tmpval64, gainsel_multi[gainsel] * 1024); + do_div(tmpval64, 128 * 1024); + + dev_dbg(&pdev->dev, "FreqMeas: %d\n", freqmeas); + dev_dbg(&pdev->dev, "BusclkFreq: %lld\n", busclk_freq); + dev_dbg(&pdev->dev, "RxRate: %lld\n", tmpval64); + + return (int)tmpval64; +} + +/* + * Get DPLL lock or not info from stable interrupt status register. + * User application must use this control to get locked, + * then can do next PCM operation + */ +static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + int rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL); + + if (spdif_priv->dpll_locked) + ucontrol->value.integer.value[0] = rate; + else + ucontrol->value.integer.value[0] = 0; + + return 0; +} + +/* User bit sync mode info */ +static int fsl_spdif_usync_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +/* + * User bit sync mode: + * 1 CD User channel subcode + * 0 Non-CD data + */ +static int fsl_spdif_usync_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 val; + + regmap_read(regmap, REG_SPDIF_SRCD, &val); + ucontrol->value.integer.value[0] = (val & SRCD_CD_USER) != 0; + + return 0; +} + +/* + * User bit sync mode: + * 1 CD User channel subcode + * 0 Non-CD data + */ +static int fsl_spdif_usync_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 val = ucontrol->value.integer.value[0] << SRCD_CD_USER_OFFSET; + + regmap_update_bits(regmap, REG_SPDIF_SRCD, SRCD_CD_USER, val); + + return 0; +} + +/* FSL SPDIF IEC958 controller defines */ +static struct snd_kcontrol_new fsl_spdif_ctrls[] = { + /* Status cchanel controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_WRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_info, + .get = fsl_spdif_pb_get, + .put = fsl_spdif_pb_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_info, + .get = fsl_spdif_capture_get, + }, + /* User bits controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Subcode Capture Default", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_info, + .get = fsl_spdif_subcode_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Q-subcode Capture Default", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_qinfo, + .get = fsl_spdif_qget, + }, + /* Valid bit error controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 V-Bit Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_vbit_info, + .get = fsl_spdif_vbit_get, + }, + /* DPLL lock info get controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "RX Sample Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_rxrate_info, + .get = fsl_spdif_rxrate_get, + }, + /* User bit sync mode set/get controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 USyncMode CDText", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_WRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_usync_info, + .get = fsl_spdif_usync_get, + .put = fsl_spdif_usync_put, + }, +}; + +static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &spdif_private->dma_params_tx; + dai->capture_dma_data = &spdif_private->dma_params_rx; + + snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls)); + + return 0; +} + +struct snd_soc_dai_driver fsl_spdif_dai = { + .probe = &fsl_spdif_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = FSL_SPDIF_RATES_PLAYBACK, + .formats = FSL_SPDIF_FORMATS_PLAYBACK, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = FSL_SPDIF_RATES_CAPTURE, + .formats = FSL_SPDIF_FORMATS_CAPTURE, + }, + .ops = &fsl_spdif_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_spdif_component = { + .name = "fsl-spdif", +}; + +/* + * ================ + * FSL SPDIF REGMAP + * ================ + */ + +static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_SPDIF_SCR: + case REG_SPDIF_SRCD: + case REG_SPDIF_SRPC: + case REG_SPDIF_SIE: + case REG_SPDIF_SIS: + case REG_SPDIF_SRL: + case REG_SPDIF_SRR: + case REG_SPDIF_SRCSH: + case REG_SPDIF_SRCSL: + case REG_SPDIF_SRU: + case REG_SPDIF_SRQ: + case REG_SPDIF_STCSCH: + case REG_SPDIF_STCSCL: + case REG_SPDIF_SRFM: + case REG_SPDIF_STC: + return true; + default: + return false; + }; +} + +static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_SPDIF_SCR: + case REG_SPDIF_SRCD: + case REG_SPDIF_SRPC: + case REG_SPDIF_SIE: + case REG_SPDIF_SIC: + case REG_SPDIF_STL: + case REG_SPDIF_STR: + case REG_SPDIF_STCSCH: + case REG_SPDIF_STCSCL: + case REG_SPDIF_STC: + return true; + default: + return false; + }; +} + +static const struct regmap_config fsl_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = REG_SPDIF_STC, + .readable_reg = fsl_spdif_readable_reg, + .writeable_reg = fsl_spdif_writeable_reg, +}; + +static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, + struct clk *clk, u64 savesub, + enum spdif_txrate index) +{ + const u32 rate[] = { 32000, 44100, 48000 }; + u64 rate_ideal, rate_actual, sub; + u32 div, arate; + + for (div = 1; div <= 128; div++) { + rate_ideal = rate[index] * (div + 1) * 64; + rate_actual = clk_round_rate(clk, rate_ideal); + + arate = rate_actual / 64; + arate /= div; + + if (arate == rate[index]) { + /* We are lucky */ + savesub = 0; + spdif_priv->txclk_div[index] = div; + break; + } else if (arate / rate[index] == 1) { + /* A little bigger than expect */ + sub = (arate - rate[index]) * 100000; + do_div(sub, rate[index]); + if (sub < savesub) { + savesub = sub; + spdif_priv->txclk_div[index] = div; + } + } else if (rate[index] / arate == 1) { + /* A little smaller than expect */ + sub = (rate[index] - arate) * 100000; + do_div(sub, rate[index]); + if (sub < savesub) { + savesub = sub; + spdif_priv->txclk_div[index] = div; + } + } + } + + return savesub; +} + +static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, + enum spdif_txrate index) +{ + const u32 rate[] = { 32000, 44100, 48000 }; + struct platform_device *pdev = spdif_priv->pdev; + struct device *dev = &pdev->dev; + u64 savesub = 100000, ret; + struct clk *clk; + char tmp[16]; + int i; + + for (i = 0; i < STC_TXCLK_SRC_MAX; i++) { + sprintf(tmp, "rxtx%d", i); + clk = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(clk)) { + dev_err(dev, "no rxtx%d clock in devicetree\n", i); + return PTR_ERR(clk); + } + if (!clk_get_rate(clk)) + continue; + + ret = fsl_spdif_txclk_caldiv(spdif_priv, clk, savesub, index); + if (savesub == ret) + continue; + + savesub = ret; + spdif_priv->txclk[index] = clk; + spdif_priv->txclk_src[index] = i; + + /* To quick catch a divisor, we allow a 0.1% deviation */ + if (savesub < 100) + break; + } + + dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate", + spdif_priv->txclk_src[index], rate[index]); + dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate", + spdif_priv->txclk_div[index], rate[index]); + + return 0; +} + +static int fsl_spdif_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_spdif_priv *spdif_priv; + struct spdif_mixer_control *ctrl; + struct resource *res; + void __iomem *regs; + int irq, ret, i; + + if (!np) + return -ENODEV; + + spdif_priv = devm_kzalloc(&pdev->dev, + sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1, + GFP_KERNEL); + if (!spdif_priv) + return -ENOMEM; + + strcpy(spdif_priv->name, np->name); + + spdif_priv->pdev = pdev; + + /* Initialize this copy of the CPU DAI driver structure */ + memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); + spdif_priv->cpu_dai_drv.name = spdif_priv->name; + + /* Get the addresses and IRQ */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (IS_ERR(res)) { + dev_err(&pdev->dev, "could not determine device resources\n"); + return PTR_ERR(res); + } + + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) { + dev_err(&pdev->dev, "could not map device resources\n"); + return PTR_ERR(regs); + } + + spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "core", regs, &fsl_spdif_regmap_config); + if (IS_ERR(spdif_priv->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + return PTR_ERR(spdif_priv->regmap); + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, + spdif_priv->name, spdif_priv); + if (ret) { + dev_err(&pdev->dev, "could not claim irq %u\n", irq); + return ret; + } + + /* Select clock source for rx/tx clock */ + spdif_priv->rxclk = devm_clk_get(&pdev->dev, "rxtx1"); + if (IS_ERR(spdif_priv->rxclk)) { + dev_err(&pdev->dev, "no rxtx1 clock in devicetree\n"); + return PTR_ERR(spdif_priv->rxclk); + } + spdif_priv->rxclk_src = DEFAULT_RXCLK_SRC; + + for (i = 0; i < SPDIF_TXRATE_MAX; i++) { + ret = fsl_spdif_probe_txclk(spdif_priv, i); + if (ret) + return ret; + } + + /* Initial spinlock for control data */ + ctrl = &spdif_priv->fsl_spdif_control; + spin_lock_init(&ctrl->ctl_lock); + + /* Init tx channel status default value */ + ctrl->ch_status[0] = + IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_5015; + ctrl->ch_status[1] = IEC958_AES1_CON_DIGDIGCONV_ID; + ctrl->ch_status[2] = 0x00; + ctrl->ch_status[3] = + IEC958_AES3_CON_FS_44100 | IEC958_AES3_CON_CLOCK_1000PPM; + + spdif_priv->dpll_locked = false; + + spdif_priv->dma_params_tx.maxburst = FSL_SPDIF_TXFIFO_WML; + spdif_priv->dma_params_rx.maxburst = FSL_SPDIF_RXFIFO_WML; + spdif_priv->dma_params_tx.addr = res->start + REG_SPDIF_STL; + spdif_priv->dma_params_rx.addr = res->start + REG_SPDIF_SRL; + + /* Register with ASoC */ + dev_set_drvdata(&pdev->dev, spdif_priv); + + ret = snd_soc_register_component(&pdev->dev, &fsl_spdif_component, + &spdif_priv->cpu_dai_drv, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); + goto error_dev; + } + + ret = imx_pcm_dma_init(pdev); + if (ret) { + dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); + goto error_component; + } + + return ret; + +error_component: + snd_soc_unregister_component(&pdev->dev); +error_dev: + dev_set_drvdata(&pdev->dev, NULL); + + return ret; +} + +static int fsl_spdif_remove(struct platform_device *pdev) +{ + imx_pcm_dma_exit(pdev); + snd_soc_unregister_component(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + return 0; +} + +static const struct of_device_id fsl_spdif_dt_ids[] = { + { .compatible = "fsl,imx35-spdif", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids); + +static struct platform_driver fsl_spdif_driver = { + .driver = { + .name = "fsl-spdif-dai", + .owner = THIS_MODULE, + .of_match_table = fsl_spdif_dt_ids, + }, + .probe = fsl_spdif_probe, + .remove = fsl_spdif_remove, +}; + +module_platform_driver(fsl_spdif_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale S/PDIF CPU DAI Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:fsl-spdif-dai"); diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h new file mode 100644 index 000000000000..b1266790d117 --- /dev/null +++ b/sound/soc/fsl/fsl_spdif.h @@ -0,0 +1,191 @@ +/* + * fsl_spdif.h - ALSA S/PDIF interface for the Freescale i.MX SoC + * + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * Based on fsl_ssi.h + * Author: Timur Tabi + * Copyright 2007-2008 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_SPDIF_DAI_H +#define _FSL_SPDIF_DAI_H + +/* S/PDIF Register Map */ +#define REG_SPDIF_SCR 0x0 /* SPDIF Configuration Register */ +#define REG_SPDIF_SRCD 0x4 /* CDText Control Register */ +#define REG_SPDIF_SRPC 0x8 /* PhaseConfig Register */ +#define REG_SPDIF_SIE 0xc /* InterruptEn Register */ +#define REG_SPDIF_SIS 0x10 /* InterruptStat Register */ +#define REG_SPDIF_SIC 0x10 /* InterruptClear Register */ +#define REG_SPDIF_SRL 0x14 /* SPDIFRxLeft Register */ +#define REG_SPDIF_SRR 0x18 /* SPDIFRxRight Register */ +#define REG_SPDIF_SRCSH 0x1c /* SPDIFRxCChannel_h Register */ +#define REG_SPDIF_SRCSL 0x20 /* SPDIFRxCChannel_l Register */ +#define REG_SPDIF_SRU 0x24 /* UchannelRx Register */ +#define REG_SPDIF_SRQ 0x28 /* QchannelRx Register */ +#define REG_SPDIF_STL 0x2C /* SPDIFTxLeft Register */ +#define REG_SPDIF_STR 0x30 /* SPDIFTxRight Register */ +#define REG_SPDIF_STCSCH 0x34 /* SPDIFTxCChannelCons_h Register */ +#define REG_SPDIF_STCSCL 0x38 /* SPDIFTxCChannelCons_l Register */ +#define REG_SPDIF_SRFM 0x44 /* FreqMeas Register */ +#define REG_SPDIF_STC 0x50 /* SPDIFTxClk Register */ + + +/* SPDIF Configuration register */ +#define SCR_RXFIFO_CTL_OFFSET 23 +#define SCR_RXFIFO_CTL_MASK (1 << SCR_RXFIFO_CTL_OFFSET) +#define SCR_RXFIFO_CTL_ZERO (1 << SCR_RXFIFO_CTL_OFFSET) +#define SCR_RXFIFO_OFF_OFFSET 22 +#define SCR_RXFIFO_OFF_MASK (1 << SCR_RXFIFO_OFF_OFFSET) +#define SCR_RXFIFO_OFF (1 << SCR_RXFIFO_OFF_OFFSET) +#define SCR_RXFIFO_RST_OFFSET 21 +#define SCR_RXFIFO_RST_MASK (1 << SCR_RXFIFO_RST_OFFSET) +#define SCR_RXFIFO_RST (1 << SCR_RXFIFO_RST_OFFSET) +#define SCR_RXFIFO_FSEL_OFFSET 19 +#define SCR_RXFIFO_FSEL_MASK (0x3 << SCR_RXFIFO_FSEL_OFFSET) +#define SCR_RXFIFO_FSEL_IF0 (0x0 << SCR_RXFIFO_FSEL_OFFSET) +#define SCR_RXFIFO_FSEL_IF4 (0x1 << SCR_RXFIFO_FSEL_OFFSET) +#define SCR_RXFIFO_FSEL_IF8 (0x2 << SCR_RXFIFO_FSEL_OFFSET) +#define SCR_RXFIFO_FSEL_IF12 (0x3 << SCR_RXFIFO_FSEL_OFFSET) +#define SCR_RXFIFO_AUTOSYNC_OFFSET 18 +#define SCR_RXFIFO_AUTOSYNC_MASK (1 << SCR_RXFIFO_AUTOSYNC_OFFSET) +#define SCR_RXFIFO_AUTOSYNC (1 << SCR_RXFIFO_AUTOSYNC_OFFSET) +#define SCR_TXFIFO_AUTOSYNC_OFFSET 17 +#define SCR_TXFIFO_AUTOSYNC_MASK (1 << SCR_TXFIFO_AUTOSYNC_OFFSET) +#define SCR_TXFIFO_AUTOSYNC (1 << SCR_TXFIFO_AUTOSYNC_OFFSET) +#define SCR_TXFIFO_FSEL_OFFSET 15 +#define SCR_TXFIFO_FSEL_MASK (0x3 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_TXFIFO_FSEL_IF0 (0x0 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_TXFIFO_FSEL_IF4 (0x1 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_TXFIFO_FSEL_IF8 (0x2 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_TXFIFO_FSEL_IF12 (0x3 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_LOW_POWER (1 << 13) +#define SCR_SOFT_RESET (1 << 12) +#define SCR_TXFIFO_CTRL_OFFSET 10 +#define SCR_TXFIFO_CTRL_MASK (0x3 << SCR_TXFIFO_CTRL_OFFSET) +#define SCR_TXFIFO_CTRL_ZERO (0x0 << SCR_TXFIFO_CTRL_OFFSET) +#define SCR_TXFIFO_CTRL_NORMAL (0x1 << SCR_TXFIFO_CTRL_OFFSET) +#define SCR_TXFIFO_CTRL_ONESAMPLE (0x2 << SCR_TXFIFO_CTRL_OFFSET) +#define SCR_DMA_RX_EN_OFFSET 9 +#define SCR_DMA_RX_EN_MASK (1 << SCR_DMA_RX_EN_OFFSET) +#define SCR_DMA_RX_EN (1 << SCR_DMA_RX_EN_OFFSET) +#define SCR_DMA_TX_EN_OFFSET 8 +#define SCR_DMA_TX_EN_MASK (1 << SCR_DMA_TX_EN_OFFSET) +#define SCR_DMA_TX_EN (1 << SCR_DMA_TX_EN_OFFSET) +#define SCR_VAL_OFFSET 5 +#define SCR_VAL_MASK (1 << SCR_VAL_OFFSET) +#define SCR_VAL_CLEAR (1 << SCR_VAL_OFFSET) +#define SCR_TXSEL_OFFSET 2 +#define SCR_TXSEL_MASK (0x7 << SCR_TXSEL_OFFSET) +#define SCR_TXSEL_OFF (0 << SCR_TXSEL_OFFSET) +#define SCR_TXSEL_RX (1 << SCR_TXSEL_OFFSET) +#define SCR_TXSEL_NORMAL (0x5 << SCR_TXSEL_OFFSET) +#define SCR_USRC_SEL_OFFSET 0x0 +#define SCR_USRC_SEL_MASK (0x3 << SCR_USRC_SEL_OFFSET) +#define SCR_USRC_SEL_NONE (0x0 << SCR_USRC_SEL_OFFSET) +#define SCR_USRC_SEL_RECV (0x1 << SCR_USRC_SEL_OFFSET) +#define SCR_USRC_SEL_CHIP (0x3 << SCR_USRC_SEL_OFFSET) + +/* SPDIF CDText control */ +#define SRCD_CD_USER_OFFSET 1 +#define SRCD_CD_USER (1 << SRCD_CD_USER_OFFSET) + +/* SPDIF Phase Configuration register */ +#define SRPC_DPLL_LOCKED (1 << 6) +#define SRPC_CLKSRC_SEL_OFFSET 7 +#define SRPC_CLKSRC_SEL_MASK (0xf << SRPC_CLKSRC_SEL_OFFSET) +#define SRPC_CLKSRC_SEL_SET(x) ((x << SRPC_CLKSRC_SEL_OFFSET) & SRPC_CLKSRC_SEL_MASK) +#define SRPC_CLKSRC_SEL_LOCKED_OFFSET1 5 +#define SRPC_CLKSRC_SEL_LOCKED_OFFSET2 2 +#define SRPC_GAINSEL_OFFSET 3 +#define SRPC_GAINSEL_MASK (0x7 << SRPC_GAINSEL_OFFSET) +#define SRPC_GAINSEL_SET(x) ((x << SRPC_GAINSEL_OFFSET) & SRPC_GAINSEL_MASK) + +#define SRPC_CLKSRC_MAX 16 + +enum spdif_gainsel { + GAINSEL_MULTI_24 = 0, + GAINSEL_MULTI_16, + GAINSEL_MULTI_12, + GAINSEL_MULTI_8, + GAINSEL_MULTI_6, + GAINSEL_MULTI_4, + GAINSEL_MULTI_3, +}; +#define GAINSEL_MULTI_MAX (GAINSEL_MULTI_3 + 1) +#define SPDIF_DEFAULT_GAINSEL GAINSEL_MULTI_8 + +/* SPDIF interrupt mask define */ +#define INT_DPLL_LOCKED (1 << 20) +#define INT_TXFIFO_UNOV (1 << 19) +#define INT_TXFIFO_RESYNC (1 << 18) +#define INT_CNEW (1 << 17) +#define INT_VAL_NOGOOD (1 << 16) +#define INT_SYM_ERR (1 << 15) +#define INT_BIT_ERR (1 << 14) +#define INT_URX_FUL (1 << 10) +#define INT_URX_OV (1 << 9) +#define INT_QRX_FUL (1 << 8) +#define INT_QRX_OV (1 << 7) +#define INT_UQ_SYNC (1 << 6) +#define INT_UQ_ERR (1 << 5) +#define INT_RXFIFO_UNOV (1 << 4) +#define INT_RXFIFO_RESYNC (1 << 3) +#define INT_LOSS_LOCK (1 << 2) +#define INT_TX_EM (1 << 1) +#define INT_RXFIFO_FUL (1 << 0) + +/* SPDIF Clock register */ +#define STC_SYSCLK_DIV_OFFSET 11 +#define STC_SYSCLK_DIV_MASK (0x1ff << STC_TXCLK_SRC_OFFSET) +#define STC_SYSCLK_DIV(x) ((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK) +#define STC_TXCLK_SRC_OFFSET 8 +#define STC_TXCLK_SRC_MASK (0x7 << STC_TXCLK_SRC_OFFSET) +#define STC_TXCLK_SRC_SET(x) ((x << STC_TXCLK_SRC_OFFSET) & STC_TXCLK_SRC_MASK) +#define STC_TXCLK_ALL_EN_OFFSET 7 +#define STC_TXCLK_ALL_EN_MASK (1 << STC_TXCLK_ALL_EN_OFFSET) +#define STC_TXCLK_ALL_EN (1 << STC_TXCLK_ALL_EN_OFFSET) +#define STC_TXCLK_DIV_OFFSET 0 +#define STC_TXCLK_DIV_MASK (0x7ff << STC_TXCLK_DIV_OFFSET) +#define STC_TXCLK_DIV(x) ((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_TXCLK_DIV_MASK) +#define STC_TXCLK_SRC_MAX 8 + +/* SPDIF tx rate */ +enum spdif_txrate { + SPDIF_TXRATE_32000 = 0, + SPDIF_TXRATE_44100, + SPDIF_TXRATE_48000, +}; +#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_48000 + 1) + + +#define SPDIF_CSTATUS_BYTE 6 +#define SPDIF_UBITS_SIZE 96 +#define SPDIF_QSUB_SIZE (SPDIF_UBITS_SIZE / 8) + + +#define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define FSL_SPDIF_RATES_CAPTURE (SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_96000) + +#define FSL_SPDIF_FORMATS_PLAYBACK (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define FSL_SPDIF_FORMATS_CAPTURE (SNDRV_PCM_FMTBIT_S24_LE) + +#endif /* _FSL_SPDIF_DAI_H */ -- cgit v1.2.3-70-g09d2 From cd7f0295aab102acb77c19d6d77eab5f5145364c Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Mon, 19 Aug 2013 17:05:58 +0200 Subject: ASoC: fsl-ssi: ac97-slave support This patch adds ac97-slave support. For ac97, the registers have to be setup earlier than for other ssi modes because there is some communication with the external device before streaming. So this patch introduces a fsl_ssi_setup function to setup the registers for different ssi operation modes seperately. This patch was tested with imx27-pca100. Signed-off-by: Markus Pargmann Tested-by: Shawn Guo Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,ssi.txt | 4 + sound/soc/fsl/fsl_ssi.c | 346 ++++++++++++++++----- 2 files changed, 280 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt index 088a2c038f01..4303b6ab6208 100644 --- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt @@ -43,6 +43,10 @@ Required properties: together. This would still allow different sample sizes, but not different sample rates. +Required are also ac97 link bindings if ac97 is used. See +Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary +bindings. + Optional properties: - codec-handle: Phandle to a 'codec' node that defines an audio codec connected to this SSI. This node is typically diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3168998dcf1d..9e410e1e49a9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -141,6 +141,7 @@ struct fsl_ssi_private { bool new_binding; bool ssi_on_imx; + bool imx_ac97; bool use_dma; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; @@ -320,6 +321,124 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) return ret; } +static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) +{ + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u8 i2s_mode; + u8 wm; + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; + + if (ssi_private->imx_ac97) + i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + else + i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + + /* + * Section 16.5 of the MPC8610 reference manual says that the SSI needs + * to be disabled before updating the registers we set here. + */ + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); + + /* + * Program the SSI into I2S Slave Non-Network Synchronous mode. Also + * enable the transmit and receive FIFO. + * + * FIXME: Little-endian samples require a different shift dir + */ + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, + CCSR_SSI_SCR_TFR_CLK_DIS | + i2s_mode | + (synchronous ? CCSR_SSI_SCR_SYN : 0)); + + write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | + CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | + CCSR_SSI_STCR_TSCKP, &ssi->stcr); + + write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | + CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | + CCSR_SSI_SRCR_RSCKP, &ssi->srcr); + /* + * The DC and PM bits are only used if the SSI is the clock master. + */ + + /* + * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't + * use FIFO 1. We program the transmit water to signal a DMA transfer + * if there are only two (or fewer) elements left in the FIFO. Two + * elements equals one frame (left channel, right channel). This value, + * however, depends on the depth of the transmit buffer. + * + * We set the watermark on the same level as the DMA burstsize. For + * fiq it is probably better to use the biggest possible watermark + * size. + */ + if (ssi_private->use_dma) + wm = ssi_private->fifo_depth - 2; + else + wm = ssi_private->fifo_depth; + + write_ssi(CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) | + CCSR_SSI_SFCSR_TFWM1(wm) | CCSR_SSI_SFCSR_RFWM1(wm), + &ssi->sfcsr); + + /* + * For non-ac97 setups, we keep the SSI disabled because if we enable + * it, then the DMA controller will start. It's not supposed to start + * until the SCR.TE (or SCR.RE) bit is set, but it does anyway. The DMA + * controller will transfer one "BWC" of data (i.e. the amount of data + * that the MR.BWC bits are set to). The reason this is bad is because + * at this point, the PCM driver has not finished initializing the DMA + * controller. + */ + + + /* + * For ac97 interrupts are enabled with the startup of the substream + * because it is also running without an active substream. Normally SSI + * is only enabled when there is a substream. + */ + if (!ssi_private->imx_ac97) { + /* Enable the interrupts and DMA requests */ + if (ssi_private->use_dma) + write_ssi(SIER_FLAGS, &ssi->sier); + else + write_ssi(CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN | + CCSR_SSI_SIER_RIE | + CCSR_SSI_SIER_RFF0_EN, &ssi->sier); + } else { + /* + * Setup the clock control register + */ + write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), + &ssi->stccr); + write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), + &ssi->srccr); + + /* + * Enable AC97 mode and startup the SSI + */ + write_ssi(CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV, + &ssi->sacnt); + write_ssi(0xff, &ssi->saccdis); + write_ssi(0x300, &ssi->saccen); + + /* + * Enable SSI + */ + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN); + write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor); + + /* + * Enable Transmit and Receive + */ + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE); + } + + return 0; +} + + /** * fsl_ssi_startup: create a new substream * @@ -341,75 +460,14 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * and initialize the SSI registers. */ if (!ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - ssi_private->first_stream = substream; /* - * Section 16.5 of the MPC8610 reference manual says that the - * SSI needs to be disabled before updating the registers we set - * here. - */ - write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); - - /* - * Program the SSI into I2S Slave Non-Network Synchronous mode. - * Also enable the transmit and receive FIFO. - * - * FIXME: Little-endian samples require a different shift dir - */ - write_ssi_mask(&ssi->scr, - CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, - CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE - | (synchronous ? CCSR_SSI_SCR_SYN : 0)); - - write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | - CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | - CCSR_SSI_STCR_TSCKP, &ssi->stcr); - - write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | - CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | - CCSR_SSI_SRCR_RSCKP, &ssi->srcr); - - /* - * The DC and PM bits are only used if the SSI is the clock - * master. - */ - - /* Enable the interrupts and DMA requests */ - if (ssi_private->use_dma) - write_ssi(SIER_FLAGS, &ssi->sier); - else - write_ssi(CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN | - CCSR_SSI_SIER_RIE | - CCSR_SSI_SIER_RFF0_EN, &ssi->sier); - - /* - * Set the watermark for transmit FIFI 0 and receive FIFO 0. We - * don't use FIFO 1. We program the transmit water to signal a - * DMA transfer if there are only two (or fewer) elements left - * in the FIFO. Two elements equals one frame (left channel, - * right channel). This value, however, depends on the depth of - * the transmit buffer. - * - * We program the receive FIFO to notify us if at least two - * elements (one frame) have been written to the FIFO. We could - * make this value larger (and maybe we should), but this way - * data will be written to memory as soon as it's available. - */ - write_ssi(CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | - CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2), - &ssi->sfcsr); - - /* - * We keep the SSI disabled because if we enable it, then the - * DMA controller will start. It's not supposed to start until - * the SCR.TE (or SCR.RE) bit is set, but it does anyway. The - * DMA controller will transfer one "BWC" of data (i.e. the - * amount of data that the MR.BWC bits are set to). The reason - * this is bad is because at this point, the PCM driver has not - * finished initializing the DMA controller. + * fsl_ssi_setup was already called by ac97_init earlier if + * the driver is in ac97 mode. */ + if (!ssi_private->imx_ac97) + fsl_ssi_setup(ssi_private); } else { if (synchronous) { struct snd_pcm_runtime *first_runtime = @@ -538,7 +596,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, else write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); - if ((read_ssi(&ssi->scr) & (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & + (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); break; @@ -608,6 +667,133 @@ static const struct snd_soc_component_driver fsl_ssi_component = { .name = "fsl-ssi", }; +/** + * fsl_ssi_ac97_trigger: start and stop the AC97 receive/transmit. + * + * This function is called by ALSA to start, stop, pause, and resume the + * transfer of data. + */ +static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata( + rtd->cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_TIE | + CCSR_SSI_SIER_TFE0_EN); + else + write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_RIE | + CCSR_SSI_SIER_RFF0_EN); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_TIE | + CCSR_SSI_SIER_TFE0_EN, 0); + else + write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_RIE | + CCSR_SSI_SIER_RFF0_EN, 0); + break; + + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + write_ssi(CCSR_SSI_SOR_TX_CLR, &ssi->sor); + else + write_ssi(CCSR_SSI_SOR_RX_CLR, &ssi->sor); + + return 0; +} + +static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = { + .startup = fsl_ssi_startup, + .shutdown = fsl_ssi_shutdown, + .trigger = fsl_ssi_ac97_trigger, +}; + +static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &fsl_ssi_ac97_dai_ops, +}; + + +static struct fsl_ssi_private *fsl_ac97_data; + +static void fsl_ssi_ac97_init(void) +{ + fsl_ssi_setup(fsl_ac97_data); +} + +void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct ccsr_ssi *ssi = fsl_ac97_data->ssi; + unsigned int lreg; + unsigned int lval; + + if (reg > 0x7f) + return; + + + lreg = reg << 12; + write_ssi(lreg, &ssi->sacadd); + + lval = val << 4; + write_ssi(lval , &ssi->sacdat); + + write_ssi_mask(&ssi->sacnt, CCSR_SSI_SACNT_RDWR_MASK, + CCSR_SSI_SACNT_WR); + udelay(100); +} + +unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct ccsr_ssi *ssi = fsl_ac97_data->ssi; + + unsigned short val = -1; + unsigned int lreg; + + lreg = (reg & 0x7f) << 12; + write_ssi(lreg, &ssi->sacadd); + write_ssi_mask(&ssi->sacnt, CCSR_SSI_SACNT_RDWR_MASK, + CCSR_SSI_SACNT_RD); + + udelay(100); + + val = (read_ssi(&ssi->sacdat) >> 4) & 0xffff; + + return val; +} + +static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = { + .read = fsl_ssi_ac97_read, + .write = fsl_ssi_ac97_write, +}; + /* Show the statistics of a flag only if its interrupt is enabled. The * compiler will optimze this code to a no-op if the interrupt is not * enabled. @@ -684,6 +870,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct resource res; char name[64]; bool shared; + bool ac97 = false; /* SSIs that are not connected on the board should have a * status = "disabled" @@ -694,7 +881,13 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* We only support the SSI in "I2S Slave" mode */ sprop = of_get_property(np, "fsl,mode", NULL); - if (!sprop || strcmp(sprop, "i2s-slave")) { + if (!sprop) { + dev_err(&pdev->dev, "fsl,mode property is necessary\n"); + return -EINVAL; + } + if (!strcmp(sprop, "ac97-slave")) { + ac97 = true; + } else if (strcmp(sprop, "i2s-slave")) { dev_notice(&pdev->dev, "mode %s is unsupported\n", sprop); return -ENODEV; } @@ -713,9 +906,19 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private->use_dma = !of_property_read_bool(np, "fsl,fiq-stream-filter"); - /* Initialize this copy of the CPU DAI driver structure */ - memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, - sizeof(fsl_ssi_dai_template)); + if (ac97) { + memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_ac97_dai, + sizeof(fsl_ssi_ac97_dai)); + + fsl_ac97_data = ssi_private; + ssi_private->imx_ac97 = true; + + snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + } else { + /* Initialize this copy of the CPU DAI driver structure */ + memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, + sizeof(fsl_ssi_dai_template)); + } ssi_private->cpu_dai_drv.name = ssi_private->name; /* Get the addresses and IRQ */ @@ -901,6 +1104,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) } done: + if (ssi_private->imx_ac97) + fsl_ssi_ac97_init(); + return 0; error_dai: -- cgit v1.2.3-70-g09d2 From f8fdf5375e2005f238ce9b430724752a6e3d55cc Mon Sep 17 00:00:00 2001 From: Steffen Trumtrar Date: Mon, 19 Aug 2013 17:05:59 +0200 Subject: ASoC: fsl-ssi: add SSIEN errata work around The chip errata for the i.MX35, Rev.2 has the following errata: ENGcm06222: SSI:Transmission does not take place in bit length early frame sync configuration The workaround states, that TX_EN and SSI_EN bits should be set in the same register write. As the next errata in the document (ENGcm06532) says to always write RX_EN and TX_EN in the same register write in network mode. Therefore include the whole write to CCSR_SSI_SCR_TE and CCSR_SSI_SCR_RE into the write to CCSR_SSI_SCR_SSIEN Signed-off-by: Steffen Trumtrar Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9e410e1e49a9..6daeb5fbdc9b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -424,15 +424,12 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) write_ssi(0x300, &ssi->saccen); /* - * Enable SSI + * Enable SSI, Transmit and Receive */ - write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN); - write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor); + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | + CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE); - /* - * Enable Transmit and Receive - */ - write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE); + write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor); } return 0; -- cgit v1.2.3-70-g09d2 From 9b443e3d89ba507ba5f51682f3896f859b2e5007 Mon Sep 17 00:00:00 2001 From: Michael Grzeschik Date: Mon, 19 Aug 2013 17:06:00 +0200 Subject: ASoC: fsl-ssi: imx-pcm-fiq bugfix imx-pcm-fiq is checking for TE RE bits, so enable them only if necessary. Signed-off-by: Michael Grzeschik Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 44 ++++++++++++++++++++++++-------------------- 1 file changed, 24 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6daeb5fbdc9b..198656fd171d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -382,31 +382,12 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) CCSR_SSI_SFCSR_TFWM1(wm) | CCSR_SSI_SFCSR_RFWM1(wm), &ssi->sfcsr); - /* - * For non-ac97 setups, we keep the SSI disabled because if we enable - * it, then the DMA controller will start. It's not supposed to start - * until the SCR.TE (or SCR.RE) bit is set, but it does anyway. The DMA - * controller will transfer one "BWC" of data (i.e. the amount of data - * that the MR.BWC bits are set to). The reason this is bad is because - * at this point, the PCM driver has not finished initializing the DMA - * controller. - */ - - /* * For ac97 interrupts are enabled with the startup of the substream * because it is also running without an active substream. Normally SSI * is only enabled when there is a substream. */ - if (!ssi_private->imx_ac97) { - /* Enable the interrupts and DMA requests */ - if (ssi_private->use_dma) - write_ssi(SIER_FLAGS, &ssi->sier); - else - write_ssi(CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN | - CCSR_SSI_SIER_RIE | - CCSR_SSI_SIER_RFF0_EN, &ssi->sier); - } else { + if (ssi_private->imx_ac97) { /* * Setup the clock control register */ @@ -574,6 +555,27 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int sier_bits; + + /* + * Enable only the interrupts and DMA requests + * that are needed for the channel. As the fiq + * is polling for this bits, we have to ensure + * that this are aligned with the preallocated + * buffers + */ + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (ssi_private->use_dma) + sier_bits = SIER_FLAGS; + else + sier_bits = CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN; + } else { + if (ssi_private->use_dma) + sier_bits = SIER_FLAGS; + else + sier_bits = CCSR_SSI_SIER_RIE | CCSR_SSI_SIER_RFF0_EN; + } switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -602,6 +604,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return -EINVAL; } + write_ssi(sier_bits, &ssi->sier); + return 0; } -- cgit v1.2.3-70-g09d2 From f037708654eef9c5477ac2a88b3a1e8b5d190dc4 Mon Sep 17 00:00:00 2001 From: Michael Grzeschik Date: Mon, 19 Aug 2013 17:06:01 +0200 Subject: ASoC: fsl: disable ssi irq for imx We have to disable the ssi irq, as it is not safe for all platforms to write back into the status register. It also runs into non-linefetch aborts. Signed-off-by: Michael Grzeschik Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 22 ++++++++++------------ 1 file changed, 10 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 198656fd171d..5cf626c4dc96 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -941,18 +941,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) return -ENXIO; } - if (ssi_private->use_dma) { - /* The 'name' should not have any slashes in it. */ - ret = devm_request_irq(&pdev->dev, ssi_private->irq, - fsl_ssi_isr, 0, ssi_private->name, - ssi_private); - if (ret < 0) { - dev_err(&pdev->dev, "could not claim irq %u\n", - ssi_private->irq); - goto error_irqmap; - } - } - /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) ssi_private->cpu_dai_drv.symmetric_rates = 1; @@ -1020,6 +1008,16 @@ static int fsl_ssi_probe(struct platform_device *pdev) dma_events[0], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx, dma_events[1], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); + } else if (ssi_private->use_dma) { + /* The 'name' should not have any slashes in it. */ + ret = devm_request_irq(&pdev->dev, ssi_private->irq, + fsl_ssi_isr, 0, ssi_private->name, + ssi_private); + if (ret < 0) { + dev_err(&pdev->dev, "could not claim irq %u\n", + ssi_private->irq); + goto error_irqmap; + } } /* Initialize the the device_attribute structure */ -- cgit v1.2.3-70-g09d2 From 06b10ff913f4d6b3e659e365ce5f70e82cca353c Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Thu, 22 Aug 2013 18:15:02 +0530 Subject: ASoC: samsung: Fix build error with dma function rename commit 85ff3c29d720 ("ASoC: samsung: Rename DMA platform registration functions") renames the DMA registration functions. Fix the places where it was left out. Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- sound/soc/samsung/pcm.c | 4 ++-- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/spdif.c | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 1566afe9ef52..e54256fc4b2c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -594,7 +594,7 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev) goto err5; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret); goto err6; @@ -623,7 +623,7 @@ static int s3c_pcm_dev_remove(struct platform_device *pdev) struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; struct resource *mem_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); pm_runtime_disable(&pdev->dev); diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 47e23864ea72..ea885cb9f76c 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -176,7 +176,7 @@ static int s3c2412_iis_dev_probe(struct platform_device *pdev) return ret; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { pr_err("failed to register the DMA: %d\n", ret); goto err; @@ -190,7 +190,7 @@ err: static int s3c2412_iis_dev_remove(struct platform_device *pdev) { - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 8b3414551a62..9c8ebd872fac 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -480,7 +480,7 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev) return ret; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { pr_err("failed to register the dma: %d\n", ret); goto err; @@ -494,7 +494,7 @@ err: static int s3c24xx_iis_dev_remove(struct platform_device *pdev) { - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 5ea70ab0ecb5..28487dcc4538 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -442,7 +442,7 @@ static int spdif_probe(struct platform_device *pdev) spdif->dma_playback = &spdif_stereo_out; - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to register DMA: %d\n", ret); goto err5; @@ -468,7 +468,7 @@ static int spdif_remove(struct platform_device *pdev) struct samsung_spdif_info *spdif = &spdif_info; struct resource *mem_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); iounmap(spdif->regs); -- cgit v1.2.3-70-g09d2