From 9b11eb44eff7ede6bc3a94511cf9dfda75af9c9f Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 6 Aug 2014 09:48:14 +0300 Subject: ASoC: Intel: Update Baytrail ADSP firmware name Update the initial Baytrail ADSP firmware file name with the one that is now in linux-firmware.git. Please see linux-firmware.git commit 7551a3a78453 ("fw_sst_0f28: Add firmware for Intel Baytrail SST DSP"). Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/sst-acpi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 42edc6f4fc4a..03d0a166b635 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -246,8 +246,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { }; static struct sst_acpi_mach baytrail_machines[] = { - { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" }, - { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" }, + { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, + { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, {} }; -- cgit v1.2.3-70-g09d2 From 27d3f02689cce5c4063a4f8dd88ce19d08a33fe6 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 11 Aug 2014 14:15:36 +0300 Subject: ASoC: Intel: Merge Baytrail ADSP suspend_noirq into suspend_late Merge DSP reset and cleanup sequence in sst_byt_pcm_dev_suspend_noirq() into sst_byt_pcm_dev_suspend_late(). First their order was wrong by first unloading firmware modules in suspend_late and then taking DSP into reset in suspend_noirq. Second ACPI has put device into OFF state already during suspend_late so trying to reset the DSP is a no-op at suspend_noirq stage. Fix these by moving DSP reset and cleanup into sst_byt_pcm_dev_suspend_late() before firmware unloading. Signed-off-by: Jarkko Nikula Tested-by: Borun Fu Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-ipc.c | 10 +--------- sound/soc/intel/sst-baytrail-ipc.h | 1 - sound/soc/intel/sst-baytrail-pcm.c | 18 ------------------ 3 files changed, 1 insertion(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index d207b22ea330..5008c8f09aac 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -797,7 +797,7 @@ static struct sst_dsp_device byt_dev = { .ops = &sst_byt_ops, }; -int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata) +int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt = pdata->dsp; @@ -806,14 +806,6 @@ int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata) sst_byt_drop_all(byt); dev_dbg(byt->dev, "dsp in reset\n"); - return 0; -} -EXPORT_SYMBOL_GPL(sst_byt_dsp_suspend_noirq); - -int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata) -{ - struct sst_byt *byt = pdata->dsp; - dev_dbg(byt->dev, "free all blocks and unload fw\n"); sst_fw_unload(byt->fw); diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h index 06a4d202689b..8faff6dcf25d 100644 --- a/sound/soc/intel/sst-baytrail-ipc.h +++ b/sound/soc/intel/sst-baytrail-ipc.h @@ -66,7 +66,6 @@ int sst_byt_get_dsp_position(struct sst_byt *byt, int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata); void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata); struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt); -int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_boot(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_wait_for_ready(struct device *dev, struct sst_pdata *pdata); diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 599401c0c655..ba7ed9720732 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -404,23 +404,6 @@ static const struct snd_soc_component_driver byt_dai_component = { }; #ifdef CONFIG_PM -static int sst_byt_pcm_dev_suspend_noirq(struct device *dev) -{ - struct sst_pdata *sst_pdata = dev_get_platdata(dev); - int ret; - - dev_dbg(dev, "suspending noirq\n"); - - /* at this point all streams will be stopped and context saved */ - ret = sst_byt_dsp_suspend_noirq(dev, sst_pdata); - if (ret < 0) { - dev_err(dev, "failed to suspend %d\n", ret); - return ret; - } - - return ret; -} - static int sst_byt_pcm_dev_suspend_late(struct device *dev) { struct sst_pdata *sst_pdata = dev_get_platdata(dev); @@ -458,7 +441,6 @@ static int sst_byt_pcm_dev_resume(struct device *dev) } static const struct dev_pm_ops sst_byt_pm_ops = { - .suspend_noirq = sst_byt_pcm_dev_suspend_noirq, .suspend_late = sst_byt_pcm_dev_suspend_late, .resume_early = sst_byt_pcm_dev_resume_early, .resume = sst_byt_pcm_dev_resume, -- cgit v1.2.3-70-g09d2 From 9246539bdda4206c53be1045778b642f1c8137e4 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 11 Aug 2014 14:15:37 +0300 Subject: ASoC: Intel: Wait Baytrail ADSP boot at resume_early stage Remove sst_byt_pcm_dev_resume() and move waiting of firmware boot into sst_byt_pcm_dev_resume_early(). Now suspend_late and resume_early phases are in sync with each other so that we know that ADSP was put into reset and was unpowered after suspend_late and is ready to resume IO after resume_early during resume stage in sst_byt_pcm_trigger(). Signed-off-by: Jarkko Nikula Tested-by: Borun Fu Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-pcm.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index ba7ed9720732..eb7b31e13565 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -423,18 +423,14 @@ static int sst_byt_pcm_dev_suspend_late(struct device *dev) static int sst_byt_pcm_dev_resume_early(struct device *dev) { struct sst_pdata *sst_pdata = dev_get_platdata(dev); + int ret; dev_dbg(dev, "resume early\n"); /* load fw and boot DSP */ - return sst_byt_dsp_boot(dev, sst_pdata); -} - -static int sst_byt_pcm_dev_resume(struct device *dev) -{ - struct sst_pdata *sst_pdata = dev_get_platdata(dev); - - dev_dbg(dev, "resume\n"); + ret = sst_byt_dsp_boot(dev, sst_pdata); + if (ret) + return ret; /* wait for FW to finish booting */ return sst_byt_dsp_wait_for_ready(dev, sst_pdata); @@ -443,7 +439,6 @@ static int sst_byt_pcm_dev_resume(struct device *dev) static const struct dev_pm_ops sst_byt_pm_ops = { .suspend_late = sst_byt_pcm_dev_suspend_late, .resume_early = sst_byt_pcm_dev_resume_early, - .resume = sst_byt_pcm_dev_resume, }; #define SST_BYT_PM_OPS (&sst_byt_pm_ops) -- cgit v1.2.3-70-g09d2 From b80d19c166c4f086eefa05308ab0cb28e43c4ca2 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 11 Aug 2014 14:15:38 +0300 Subject: ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in reset There is no need to restore and restart PCM streams in case ADSP didn't reach reset and power off state during system suspend/resume cycle. In that case stream is still active but paused and firmware doesn't allow allocating a new stream before paused stream is freed. ADSP remains active in case suspend sequence didn't go to suspend_late stage. This can happen when either suspend sequence is aborted by a wakeup or by letting only devices suspend by "echo devices >/sys/power/pm_test". Currently stream restoring fails in these suspend cases. Fix this by adding a flag that indicates is complete stream reinitialization needed or is it enough to resume paused stream. Flag is set when we know that ADSP reached suspend_late. Initial fix to this issue came from Fang Yang. I modified it a little and forward ported it to top of two other suspend/resume patches from me. Signed-off-by: Jarkko Nikula Tested-by: Borun Fu Cc: yang fang Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-pcm.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index eb7b31e13565..eab1c7d85187 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -59,6 +59,9 @@ struct sst_byt_priv_data { /* DAI data */ struct sst_byt_pcm_data pcm[BYT_PCM_COUNT]; + + /* flag indicating is stream context restore needed after suspend */ + bool restore_stream; }; /* this may get called several times by oss emulation */ @@ -184,7 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: - schedule_work(&pcm_data->work); + if (pdata->restore_stream == true) + schedule_work(&pcm_data->work); + else + sst_byt_stream_resume(byt, pcm_data->stream); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: sst_byt_stream_resume(byt, pcm_data->stream); @@ -193,6 +199,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_stop(byt, pcm_data->stream); break; case SNDRV_PCM_TRIGGER_SUSPEND: + pdata->restore_stream = false; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; @@ -407,6 +414,7 @@ static const struct snd_soc_component_driver byt_dai_component = { static int sst_byt_pcm_dev_suspend_late(struct device *dev) { struct sst_pdata *sst_pdata = dev_get_platdata(dev); + struct sst_byt_priv_data *priv_data = dev_get_drvdata(dev); int ret; dev_dbg(dev, "suspending late\n"); @@ -417,6 +425,8 @@ static int sst_byt_pcm_dev_suspend_late(struct device *dev) return ret; } + priv_data->restore_stream = true; + return ret; } -- cgit v1.2.3-70-g09d2 From 6912831623c5bbd38c6c26039d5f821557e5f541 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Fri, 8 Aug 2014 17:29:35 +0200 Subject: ASoC: dapm: Fix uninitialized variable in snd_soc_dapm_get_enum_double() If soc_dapm_read() fails, reg_val will be uninitialized, and bogus values will be written later: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_get_enum_double': sound/soc/soc-dapm.c:2862:15: warning: 'reg_val' may be used uninitialized in this function [-Wmaybe-uninitialized] unsigned int reg_val, val; ^ Return early on error to fix this. Introduced by commit ce0fc93ae56e2ba50ff8c220d69e4e860e889320 ("ASoC: Add DAPM support at the component level"). Signed-off-by: Geert Uytterhoeven Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/soc-dapm.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8348352dc2c6..177bd8639ef9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2860,12 +2860,14 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - int ret = 0; - if (e->reg != SND_SOC_NOPM) - ret = soc_dapm_read(dapm, e->reg, ®_val); - else + if (e->reg != SND_SOC_NOPM) { + int ret = soc_dapm_read(dapm, e->reg, ®_val); + if (ret) + return ret; + } else { reg_val = dapm_kcontrol_get_value(kcontrol); + } val = (reg_val >> e->shift_l) & e->mask; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); @@ -2875,7 +2877,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, ucontrol->value.enumerated.item[1] = val; } - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); -- cgit v1.2.3-70-g09d2 From 1c6d36805fcbd9f84b6d9252b4d022653df8d1fa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 8 Aug 2014 16:04:01 +0100 Subject: ASoC: pcm512x: Correct Digital Playback control names The source type should come before the direction specifier according to ControlNames.txt. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 163ec3855fd4..0c8aefab404c 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds = pcm512x_ramp_step_text); static const struct snd_kcontrol_new pcm512x_controls[] = { -SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, +SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL, PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), -SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, +SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, PCM512x_RQMR_SHIFT, 1, 1), SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), -- cgit v1.2.3-70-g09d2 From d114e5f73b1191a6c323eb1f25fd5084db6539cc Mon Sep 17 00:00:00 2001 From: Nikesh Oswal Date: Tue, 12 Aug 2014 15:30:32 +0100 Subject: ASoC: arizona: Fix TDM slot length handling in arizona_hw_params TDM slot length was set same as word length, regardless of the value received in set_tdm_slot. This patch sets the TDM slot length correctly as received in set_tdm_slot DAI callback Signed-off-by: Nikesh Oswal Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 2f2e91ac690f..4dfab9573a95 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; + wl = snd_pcm_format_width(params_format(params)); + if (tdm_slots) { arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", tdm_slots, tdm_width); @@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, channels = tdm_slots; } else { bclk_target = snd_soc_params_to_bclk(params); + tdm_width = wl; } if (chan_limit && chan_limit < channels) { @@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", rates[bclk], rates[bclk] / lrclk); - wl = snd_pcm_format_width(params_format(params)); - frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width; reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame); -- cgit v1.2.3-70-g09d2 From 8813543ecb405f3ea29be8dfa1f85afc6e06a544 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 6 Aug 2014 16:47:16 +0300 Subject: ASoC: mcasp: Fix implicit BLCK divider setting The implicit BLCK divider setting was broken by "ASoC: mcasp: don't override bclk divider if it was provided by the machine"-patch. After the BCLK divider is implicitly set for the first time the mcasp->bclk_div gets a non zero value and the implicit setting is "turned off". Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c28508da34cf..6a6b2ff7d7d7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -403,7 +403,8 @@ out: return ret; } -static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, + int div, bool explicit) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); @@ -420,7 +421,8 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div ACLKXDIV(div - 1), ACLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); - mcasp->bclk_div = div; + if (explicit) + mcasp->bclk_div = div; break; case 2: /* BCLK/LRCLK ratio */ @@ -434,6 +436,12 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div return 0; } +static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, + int div) +{ + return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1); +} + static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -738,7 +746,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, "Inaccurate BCLK: %u Hz / %u != %u Hz\n", mcasp->sysclk_freq, div, bclk_freq); } - davinci_mcasp_set_clkdiv(cpu_dai, 1, div); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } ret = mcasp_common_hw_param(mcasp, substream->stream, -- cgit v1.2.3-70-g09d2 From 769091ee18056b3aa35b415d9768fb23f361e598 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 8 Aug 2014 14:47:22 +0800 Subject: ASoC: fsl-esai: Revert .xlate_tdm_slot_mask() support This reverts commit a603c8ee526f5ea9ad9b40710308766299ad8a69. fsl_asoc_xlate_tdm_slot_mask() is different with snd_soc_xlate_tdm_slot_mask(). fsl_asoc_xlate_tdm_slot_mask() will set the enabled bit to 0, disabled bit to 1. snd_soc_xlate_tdm_slot_mask() will set the enabled bit to 1, disabled bit to 0. For esai when the bit value is 1, the slot is enabled, when the bit value is 0, the slot is disabled. If using fsl_asoc_xlate_tdm_slot_mask(), the esai will work abnormally. So revert this patch, make the esai use default function. Signed-off-by: Shengjiu Wang Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/fsl_esai.c | 2 -- 2 files changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index f54a8fc99291..f3012b645b51 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -49,7 +49,6 @@ config SND_SOC_FSL_ESAI tristate "Enhanced Serial Audio Interface (ESAI) module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_FSL_UTILS help Say Y if you want to add Enhanced Synchronous Audio Interface (ESAI) support for the Freescale CPUs. diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 72d154e7dd03..a3b29ed84963 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -18,7 +18,6 @@ #include "fsl_esai.h" #include "imx-pcm.h" -#include "fsl_utils.h" #define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000 #define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ @@ -607,7 +606,6 @@ static struct snd_soc_dai_ops fsl_esai_dai_ops = { .hw_params = fsl_esai_hw_params, .set_sysclk = fsl_esai_set_dai_sysclk, .set_fmt = fsl_esai_set_dai_fmt, - .xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask, .set_tdm_slot = fsl_esai_set_dai_tdm_slot, }; -- cgit v1.2.3-70-g09d2 From 9301503af016eb537ccce76adec0c1bb5c84871e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 13 Aug 2014 21:51:06 +0200 Subject: ASoC: pxa-ssp: drop SNDRV_PCM_FMTBIT_S24_LE This mode is unsupported, as the DMA controller can't do zero-padding of samples. Signed-off-by: Daniel Mack Reported-by: Johannes Stezenbach Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/pxa/pxa-ssp.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 0109f6c2334e..a8e097433074 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -765,9 +765,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai) SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .startup = pxa_ssp_startup, -- cgit v1.2.3-70-g09d2 From f3ee07d8b6e061bf34a7167c3f564e8da4360a99 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Aug 2014 17:35:00 +0200 Subject: ALSA: hda/realtek - Avoid setting wrong COEF on ALC269 & co ALC269 & co have many vendor-specific setups with COEF verbs. However, some verbs seem specific to some codec versions and they result in the codec stalling. Typically, such a case can be avoided by checking the return value from reading a COEF. If the return value is -1, it implies that the COEF is invalid, thus it shouldn't be written. This patch adds the invalid COEF checks in appropriate places accessing ALC269 and its variants. The patch actually fixes the resume problem on Acer AO725 laptop. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181 Tested-by: Francesco Muzio Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6b38ec3c6e57..b32ce086d2e0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -181,6 +181,8 @@ static void alc_fix_pll(struct hda_codec *codec) spec->pll_coef_idx); val = snd_hda_codec_read(codec, spec->pll_nid, 0, AC_VERB_GET_PROC_COEF, 0); + if (val == -1) + return; snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX, spec->pll_coef_idx); snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF, @@ -2806,6 +2808,8 @@ static void alc286_shutup(struct hda_codec *codec) static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { int val = alc_read_coef_idx(codec, 0x04); + if (val == -1) + return; if (power_up) val |= 1 << 11; else @@ -5311,27 +5315,30 @@ static void alc269_fill_coef(struct hda_codec *codec) if ((alc_get_coef0(codec) & 0x00ff) == 0x017) { val = alc_read_coef_idx(codec, 0x04); /* Power up output pin */ - alc_write_coef_idx(codec, 0x04, val | (1<<11)); + if (val != -1) + alc_write_coef_idx(codec, 0x04, val | (1<<11)); } if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { val = alc_read_coef_idx(codec, 0xd); - if ((val & 0x0c00) >> 10 != 0x1) { + if (val != -1 && (val & 0x0c00) >> 10 != 0x1) { /* Capless ramp up clock control */ alc_write_coef_idx(codec, 0xd, val | (1<<10)); } val = alc_read_coef_idx(codec, 0x17); - if ((val & 0x01c0) >> 6 != 0x4) { + if (val != -1 && (val & 0x01c0) >> 6 != 0x4) { /* Class D power on reset */ alc_write_coef_idx(codec, 0x17, val | (1<<7)); } } val = alc_read_coef_idx(codec, 0xd); /* Class D */ - alc_write_coef_idx(codec, 0xd, val | (1<<14)); + if (val != -1) + alc_write_coef_idx(codec, 0xd, val | (1<<14)); val = alc_read_coef_idx(codec, 0x4); /* HP */ - alc_write_coef_idx(codec, 0x4, val | (1<<11)); + if (val != -1) + alc_write_coef_idx(codec, 0x4, val | (1<<11)); } /* -- cgit v1.2.3-70-g09d2 From f294afed03b154fbfaa9a32a0ebe7abdbf98070c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 17 Aug 2014 11:28:35 +0200 Subject: ASoC: Use dev_set_name() instead of init_name init_name is basically a hack and should only be used for statically allocated device structs. For dynamically allocated devices dev_set_name() should be used. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d4bfd4a9076f..889f4e3d35dc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1325,7 +1325,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; - rtd->dev->init_name = name; + dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); -- cgit v1.2.3-70-g09d2 From c8e6e960733f4a5835265c15429fced4d2f1595e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 17 Aug 2014 16:18:19 +0200 Subject: ASoC: rcar: Use && instead of & for boolean expressions Sparse spits out the following warning: sound/soc/sh/rcar/gen.c:250:21: warning: dubious: x & !y It does this because sometimes mixing boolean and bit-wise logic has not the intended result. In this case we are fine, but replacing the bit-wise '&' with the boolean '&&' silences the sparse warning. The generated code for both cases is the same. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 3fdf3be7b99a..f95e7ab135e8 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -247,7 +247,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv, }; /* it shouldn't happen */ - if (use_dvc & !use_src) + if (use_dvc && !use_src) dev_err(dev, "DVC is selected without SRC\n"); /* use SSIU or SSI ? */ -- cgit v1.2.3-70-g09d2 From f475371aa65de84fa483a998ab7594531026b9d9 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Tue, 19 Aug 2014 12:07:03 +0800 Subject: ALSA: hda - restore the gpio led after resume On some HP laptops, the mute led is controlled by codec gpio. When some machine resume from s3/s4, the codec gpio data will be cleared to 0 by BIOS: Before suspend: IO[3]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 After resume: IO[3]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 To skip the AFG node to enter D3 can't fix this problem. A workaround is to restore the gpio data when the system resume back from s3/s4. It is safe even on the machines without this problem. BugLink: https://bugs.launchpad.net/bugs/1358116 Tested-by: Franz Hsieh Cc: stable@vger.kernel.org Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b32ce086d2e0..d71270a3f73f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3268,6 +3268,15 @@ static int alc269_resume(struct hda_codec *codec) snd_hda_codec_resume_cache(codec); alc_inv_dmic_sync(codec, true); hda_call_check_power_status(codec, 0x01); + + /* on some machine, the BIOS will clear the codec gpio data when enter + * suspend, and won't restore the data after resume, so we restore it + * in the driver. + */ + if (spec->gpio_led) + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); + if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); -- cgit v1.2.3-70-g09d2 From d35f64e748e7752a5a60b1c7798cece51d19a213 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Tue, 19 Aug 2014 16:20:11 +0800 Subject: ALSA: hda/hdmi - set depop_delay for haswell plus Both Haswell and Broadwell need set depop_delay to 0. So apply this setting to haswell plus. Signed-off-by: Libin Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 36badba2dcec..5e229f7eaf24 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2330,9 +2330,8 @@ static int patch_generic_hdmi(struct hda_codec *codec) intel_haswell_fixup_enable_dp12(codec); } - if (is_haswell(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview(codec)) codec->depop_delay = 0; - } if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; -- cgit v1.2.3-70-g09d2 From ca2e7224d7e7d424e69616634f90f3f428710085 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Tue, 19 Aug 2014 16:20:12 +0800 Subject: ALSA: hda/hdmi - apply Valleyview fix-ups to Cherryview display codec Valleyview and Cherryview have the same behavior on display audio. So this patch defines is_valleyview_plus() to include codecs for both Valleyview and its successor Cherryview, and apply Valleyview fix-ups to Cherryview. Signed-off-by: Libin Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5e229f7eaf24..99d7d7fecaad 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -50,6 +50,8 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec)) #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) +#define is_cherryview(codec) ((codec)->vendor_id == 0x80862883) +#define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec)) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1459,7 +1461,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell_plus(codec) || is_valleyview(codec)) + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1598,7 +1600,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * and this can make HW reset converter selection on a pin. */ if (eld->eld_valid && !old_eld_valid && per_pin->setup) { - if (is_haswell_plus(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || + is_valleyview_plus(codec)) { intel_verify_pin_cvt_connect(codec, per_pin); intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); @@ -1779,7 +1782,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, bool non_pcm; int pinctl; - if (is_haswell_plus(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { /* Verify pin:cvt selections to avoid silent audio after S3. * After S3, the audio driver restores pin:cvt selections * but this can happen before gfx is ready and such selection @@ -2330,7 +2333,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) intel_haswell_fixup_enable_dp12(codec); } - if (is_haswell_plus(codec) || is_valleyview(codec)) + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) codec->depop_delay = 0; if (hdmi_parse_codec(codec) < 0) { -- cgit v1.2.3-70-g09d2 From ddc64b278a4dda052390b3de1b551e59acdff105 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 21 Aug 2014 20:55:21 +0200 Subject: ALSA: core: fix buffer overflow in snd_info_get_line() snd_info_get_line() documents that its last parameter must be one less than the buffer size, but this API design guarantees that (literally) every caller gets it wrong. Just change this parameter to have its obvious meaning. Reported-by: Tommi Rantala Cc: # v2.2.26+ Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/core/info.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/info.c b/sound/core/info.c index 051d55b05521..9f404e965ea2 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -684,7 +684,7 @@ int snd_info_card_free(struct snd_card *card) * snd_info_get_line - read one line from the procfs buffer * @buffer: the procfs buffer * @line: the buffer to store - * @len: the max. buffer size - 1 + * @len: the max. buffer size * * Reads one line from the buffer and stores the string. * @@ -704,7 +704,7 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) buffer->stop = 1; if (c == '\n') break; - if (len) { + if (len > 1) { len--; *line++ = c; } -- cgit v1.2.3-70-g09d2 From 3c25d041293879a8b7ff522f3a42267c45a3ab79 Mon Sep 17 00:00:00 2001 From: Rasmus Villemoes Date: Fri, 22 Aug 2014 11:22:09 +0200 Subject: ALSA: hda: ca0132_regs.h: Fix typo in include guard Signed-off-by: Rasmus Villemoes Signed-off-by: Takashi Iwai --- sound/pci/hda/ca0132_regs.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/ca0132_regs.h b/sound/pci/hda/ca0132_regs.h index 07e760937d3c..8371274aa811 100644 --- a/sound/pci/hda/ca0132_regs.h +++ b/sound/pci/hda/ca0132_regs.h @@ -20,7 +20,7 @@ */ #ifndef __CA0132_REGS_H -#define __CA0312_REGS_H +#define __CA0132_REGS_H #define DSP_CHIP_OFFSET 0x100000 #define DSP_DBGCNTL_MODULE_OFFSET 0xE30 -- cgit v1.2.3-70-g09d2 From ee3043b2d7b1bfe03cd697b144abf25954ec5fc6 Mon Sep 17 00:00:00 2001 From: Rasmus Villemoes Date: Fri, 22 Aug 2014 11:23:09 +0200 Subject: ALSA: ctxfi: ct20k1reg: Fix typo in include guard Signed-off-by: Rasmus Villemoes Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ct20k1reg.h | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ct20k1reg.h b/sound/pci/ctxfi/ct20k1reg.h index f2e34e3f27ee..5851249f11d9 100644 --- a/sound/pci/ctxfi/ct20k1reg.h +++ b/sound/pci/ctxfi/ct20k1reg.h @@ -7,7 +7,7 @@ */ #ifndef CT20K1REG_H -#define CT20k1REG_H +#define CT20K1REG_H /* 20k1 registers */ #define DSPXRAM_START 0x000000 @@ -632,5 +632,3 @@ #define I2SD_R 0x19L #endif /* CT20K1REG_H */ - - -- cgit v1.2.3-70-g09d2 From 94a988a8ab91c0cdabd2431281ec09dc52d92674 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 22 Aug 2014 11:18:48 +0200 Subject: ALSA: pcm: Fix the silence data for DSD formats Right now we set 0 as the silence data for DSD_U8 and DSD_U16 formats, but this is actually wrong. 0 is rather the most negative value. Alternatively, we may take the repeating 0x69 pattern like ffmpeg deploys. Reference: https://ffmpeg.org/pipermail/ffmpeg-cvslog/2014-April/076427.html Suggested-by: Alexander E. Patrakov Signed-off-by: Takashi Iwai --- sound/core/pcm_misc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 4560ca0e5651..2c6fd80e0bd1 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -142,11 +142,11 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { }, [SNDRV_PCM_FORMAT_DSD_U8] = { .width = 8, .phys = 8, .le = 1, .signd = 0, - .silence = {}, + .silence = { 0x69 }, }, [SNDRV_PCM_FORMAT_DSD_U16_LE] = { .width = 16, .phys = 16, .le = 1, .signd = 0, - .silence = {}, + .silence = { 0x69, 0x69 }, }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { -- cgit v1.2.3-70-g09d2 From aa47746269b0f87b3c042db7453b9e461029aed7 Mon Sep 17 00:00:00 2001 From: Rasmus Villemoes Date: Fri, 22 Aug 2014 11:25:13 +0200 Subject: ASoC: da732x: Fix typo in include guard Signed-off-by: Rasmus Villemoes Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h index 1dceafeec415..f586cbd30b77 100644 --- a/sound/soc/codecs/da732x.h +++ b/sound/soc/codecs/da732x.h @@ -11,7 +11,7 @@ */ #ifndef __DA732X_H_ -#define __DA732X_H +#define __DA732X_H_ #include -- cgit v1.2.3-70-g09d2 From d50884afdf592ebfe449b0a7cd741dd658716b13 Mon Sep 17 00:00:00 2001 From: Rasmus Villemoes Date: Fri, 22 Aug 2014 11:27:07 +0200 Subject: ASoC: tegra: Fix typo in include guard Signed-off-by: Rasmus Villemoes Acked-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_utils.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 9577121ce971..ca8037634100 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -21,7 +21,7 @@ */ #ifndef __TEGRA_ASOC_UTILS_H__ -#define __TEGRA_ASOC_UTILS_H_ +#define __TEGRA_ASOC_UTILS_H__ struct clk; struct device; -- cgit v1.2.3-70-g09d2 From 1a22e7758eabc431d6d8af085dc6e4c5031779a6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Aug 2014 08:19:05 +0200 Subject: ALSA: hda - Set up initial pins for Acer Aspire V5 Acer Aspire V5 doesn't set up the pins correctly at the cold boot while the pins are corrected after the warm reboot. This patch gives the proper pin configs statically in the driver as a workaround. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=81561 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d71270a3f73f..d446ac3137b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4408,6 +4408,7 @@ enum { ALC292_FIXUP_TPT440_DOCK, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, + ALC282_FIXUP_ASPIRE_V5_PINS, }; static const struct hda_fixup alc269_fixups[] = { @@ -4855,6 +4856,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained_before = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC282_FIXUP_ASPIRE_V5_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x90a60130 }, + { 0x14, 0x90170110 }, + { 0x17, 0x40000008 }, + { 0x18, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { 0x1a, 0x411111f0 }, + { 0x1b, 0x411111f0 }, + { 0x1d, 0x40f89b2d }, + { 0x1e, 0x411111f0 }, + { 0x21, 0x0321101f }, + { }, + }, + }, }; @@ -4866,6 +4883,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), + SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- cgit v1.2.3-70-g09d2 From f4821e8e8e957fe4c601a49b9a97b7399d5f7ab1 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 26 Aug 2014 17:03:13 +0300 Subject: ASoC: rt5640: Do not allow regmap to use bulk read-write operations Debugging showed Realtek RT5642 doesn't support autoincrementing writes so driver should set the use_single_rw flag for regmap. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6bc6efdec550..f1ec6e6bd08a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2059,6 +2059,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { static const struct regmap_config rt5640_regmap = { .reg_bits = 8, .val_bits = 16, + .use_single_rw = true, .max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) * RT5640_PR_SPACING), -- cgit v1.2.3-70-g09d2 From 22e51345a9f272e20cea3d679dca8a0e19a178e1 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 27 Aug 2014 19:50:33 +0800 Subject: ASoC: rt5677: correct mismatch widget name We name MICBIAS1 in dapm widget, but micbias1 in route table. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 67f14556462f..5337c448b5e3 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2135,10 +2135,10 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "BST2", NULL, "IN2P" }, { "BST2", NULL, "IN2N" }, - { "IN1P", NULL, "micbias1" }, - { "IN1N", NULL, "micbias1" }, - { "IN2P", NULL, "micbias1" }, - { "IN2N", NULL, "micbias1" }, + { "IN1P", NULL, "MICBIAS1" }, + { "IN1N", NULL, "MICBIAS1" }, + { "IN2P", NULL, "MICBIAS1" }, + { "IN2N", NULL, "MICBIAS1" }, { "ADC 1", NULL, "BST1" }, { "ADC 1", NULL, "ADC 1 power" }, -- cgit v1.2.3-70-g09d2 From c98853aec1f7a05545642b6ca8591fd13b2fc7b6 Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Thu, 28 Aug 2014 10:54:09 -0500 Subject: ASoC: cs4265: Fix clock rates in clock map table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Reported-by: Zoltán Szenczi Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs4265.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index a20b30ca52c0..8811689e372b 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -282,10 +282,10 @@ static const struct cs4265_clk_para clk_map_table[] = { /*64k*/ {8192000, 64000, 1, 0}, - {1228800, 64000, 1, 1}, - {1693440, 64000, 1, 2}, - {2457600, 64000, 1, 3}, - {3276800, 64000, 1, 4}, + {12288000, 64000, 1, 1}, + {16934400, 64000, 1, 2}, + {24576000, 64000, 1, 3}, + {32768000, 64000, 1, 4}, /* 88.2k */ {11289600, 88200, 1, 0}, -- cgit v1.2.3-70-g09d2 From fb18cd2a62f597557d5078d8fa03bb6930fe839f Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Thu, 28 Aug 2014 10:54:10 -0500 Subject: ASoC: cs4265: Fix setting of functional mode and clock divider MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Reported-by: Zoltán Szenczi Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs4265.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 8811689e372b..98523209f739 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -435,10 +435,10 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params)); if (index >= 0) { snd_soc_update_bits(codec, CS4265_ADC_CTL, - CS4265_ADC_FM, clk_map_table[index].fm_mode); + CS4265_ADC_FM, clk_map_table[index].fm_mode << 6); snd_soc_update_bits(codec, CS4265_MCLK_FREQ, CS4265_MCLK_FREQ_MASK, - clk_map_table[index].mclkdiv); + clk_map_table[index].mclkdiv << 4); } else { dev_err(codec->dev, "can't get correct mclk\n"); -- cgit v1.2.3-70-g09d2 From 1033eb5b5aeeb526c22068e0fb0cef9f3c14231e Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 29 Aug 2014 13:40:44 +0900 Subject: ALSA: dice: fix wrong channel mappping at higher sampling rate The channel mapping is initialized by amdtp_stream_set_parameters(), however Dice driver set it before calling this function. Furthermore, the setting is wrong because the index is the value of array, and vice versa. This commit moves codes for channel mapping after the function and set it correctly. Reported-by: Daniel Robbins Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") Signed-off-by: Takashi Sakamoto Cc: # 3.16 Signed-off-by: Takashi Iwai --- sound/firewire/dice.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index a9a30c0161f1..4cf8eb704045 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -579,11 +579,6 @@ static int dice_hw_params(struct snd_pcm_substream *substream, return err; } - for (i = 0; i < channels; i++) { - dice->stream.pcm_positions[i * 2] = i; - dice->stream.pcm_positions[i * 2 + 1] = i + channels; - } - rate /= 2; channels *= 2; } @@ -591,6 +586,15 @@ static int dice_hw_params(struct snd_pcm_substream *substream, mode = rate_index_to_mode(rate_index); amdtp_stream_set_parameters(&dice->stream, rate, channels, dice->rx_midi_ports[mode]); + if (rate_index > 4) { + channels /= 2; + + for (i = 0; i < channels; i++) { + dice->stream.pcm_positions[i] = i * 2; + dice->stream.pcm_positions[i + channels] = i * 2 + 1; + } + } + amdtp_stream_set_pcm_format(&dice->stream, params_format(hw_params)); -- cgit v1.2.3-70-g09d2 From 65845f29bec6bc17f80eff25c3bc39bcf3be9bf9 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 29 Aug 2014 13:40:45 +0900 Subject: ALSA: firewire-lib/dice: add arrangements of PCM pointer and interrupts for Dice quirk In IEC 61883-6, one data block transfers one event. In ALSA, the event equals one PCM frame, hence one data block transfers one PCM frame. But Dice has a quirk at higher sampling rate (176.4/192.0 kHz) that one data block transfers two PCM frames. Commit 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") moved some codes related to this quirk into Dice driver. But the commit forgot to add arrangements for PCM period interrupts and DMA pointer updates. As a result, Dice driver cannot work correctly at higher sampling rate. This commit adds 'double_pcm_frames' parameter to amdtp structure for this quirk. When this parameter is set, PCM period interrupts and DMA pointer updates occur at double speed than in IEC 61883-6. Reported-by: Daniel Robbins Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") Signed-off-by: Takashi Sakamoto Cc: # 3.16 Signed-off-by: Takashi Iwai --- sound/firewire/amdtp.c | 11 ++++++++++- sound/firewire/amdtp.h | 1 + sound/firewire/dice.c | 15 +++++++++++---- 3 files changed, 22 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index f96bf4c7c232..95fc2eaf11dc 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -507,7 +507,16 @@ static void amdtp_pull_midi(struct amdtp_stream *s, static void update_pcm_pointers(struct amdtp_stream *s, struct snd_pcm_substream *pcm, unsigned int frames) -{ unsigned int ptr; +{ + unsigned int ptr; + + /* + * In IEC 61883-6, one data block represents one event. In ALSA, one + * event equals to one PCM frame. But Dice has a quirk to transfer + * two PCM frames in one data block. + */ + if (s->double_pcm_frames) + frames *= 2; ptr = s->pcm_buffer_pointer + frames; if (ptr >= pcm->runtime->buffer_size) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index d8ee7b0e9386..4823c08196ac 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -125,6 +125,7 @@ struct amdtp_stream { unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; bool pointer_flush; + bool double_pcm_frames; struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8]; diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 4cf8eb704045..e3a04d69c853 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -567,10 +567,14 @@ static int dice_hw_params(struct snd_pcm_substream *substream, return err; /* - * At rates above 96 kHz, pretend that the stream runs at half the - * actual sample rate with twice the number of channels; two samples - * of a channel are stored consecutively in the packet. Requires - * blocking mode and PCM buffer size should be aligned to SYT_INTERVAL. + * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in + * one data block of AMDTP packet. Thus sampling transfer frequency is + * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are + * transferred on AMDTP packets at 96 kHz. Two successive samples of a + * channel are stored consecutively in the packet. This quirk is called + * as 'Dual Wire'. + * For this quirk, blocking mode is required and PCM buffer size should + * be aligned to SYT_INTERVAL. */ channels = params_channels(hw_params); if (rate_index > 4) { @@ -581,6 +585,9 @@ static int dice_hw_params(struct snd_pcm_substream *substream, rate /= 2; channels *= 2; + dice->stream.double_pcm_frames = true; + } else { + dice->stream.double_pcm_frames = false; } mode = rate_index_to_mode(rate_index); -- cgit v1.2.3-70-g09d2 From fdaf42c0105a24de8aefa60f6f7360842c4e673e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Aug 2014 13:30:23 +0300 Subject: ASoC: omap-twl4030: Fix typo in 2nd dai link's platform_name The platform_name should be omap-mcasp3 for the 2nd link which is used for voice connection. Reported-by: Tony Lindgren Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/omap/omap-twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index f8a6adc2d81c..4336d1831485 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -260,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { .stream_name = "TWL4030 Voice", .cpu_dai_name = "omap-mcbsp.3", .codec_dai_name = "twl4030-voice", - .platform_name = "omap-mcbsp.2", + .platform_name = "omap-mcbsp.3", .codec_name = "twl4030-codec", .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM, -- cgit v1.2.3-70-g09d2 From ff50479ad61069f3ee14863225aebe36d598e93e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 1 Sep 2014 14:26:49 +0200 Subject: ALSA: hda - Fix digital mic on Acer Aspire 3830TG Acer Aspire 3830TG with CX20588 codec has a digital built-in mic that has the same problem like many others, the inverted signal in stereo. Apply the same fixup to this machine, too. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6f2fa838b635..6e5d0cb4e3d7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -217,6 +217,7 @@ enum { CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, CXT_FIXUP_GPIO1, + CXT_FIXUP_ASPIRE_DMIC, CXT_FIXUP_THINKPAD_ACPI, CXT_FIXUP_OLPC_XO, CXT_FIXUP_CAP_MIX_AMP, @@ -664,6 +665,12 @@ static const struct hda_fixup cxt_fixups[] = { { } }, }, + [CXT_FIXUP_ASPIRE_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_stereo_dmic, + .chained = true, + .chain_id = CXT_FIXUP_GPIO1, + }, [CXT_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, .v.func = hda_fixup_thinkpad_acpi, @@ -744,7 +751,7 @@ static const struct hda_model_fixup cxt5051_fixup_models[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- cgit v1.2.3-70-g09d2 From e3c4a28b611b03d69bfbdffda985ef0dd94c2794 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 1 Sep 2014 14:46:52 +0800 Subject: ASoC: simple-card: Fix bug of wrong decrement DT node's refcount DAI links's cpu_of_node's and codec_of_node's refcounts shouldn't be decremented immediately at the end of the probe() fucntion. Because we will still use them before the audio card is removed. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 159e517fa09a..cef7776b712c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&priv->snd_card, priv); ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); + if (ret >= 0) + return ret; err: asoc_simple_card_unref(pdev); return ret; } +static int asoc_simple_card_remove(struct platform_device *pdev) +{ + return asoc_simple_card_unref(pdev); +} + static const struct of_device_id asoc_simple_of_match[] = { { .compatible = "simple-audio-card", }, {}, @@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = { .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, + .remove = asoc_simple_card_remove, }; module_platform_driver(asoc_simple_card); -- cgit v1.2.3-70-g09d2 From acf08081adb5e8fe0519eb97bb49797ef52614d6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Sep 2014 07:21:56 +0200 Subject: ALSA: hda - Fix COEF setups for ALC1150 codec ALC1150 codec seems to need the COEF- and PLL-setups just like its compatible ALC882 codec. Some machines (e.g. SunMicro X10SAT) show the problem like too low output volumes unless the COEF setup is applied. Reported-and-tested-by: Dana Goyette Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d446ac3137b3..1ba22fb527c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -328,6 +328,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0885: case 0x10ec0887: /*case 0x10ec0889:*/ /* this causes an SPDIF problem */ + case 0x10ec0900: alc889_coef_init(codec); break; case 0x10ec0888: @@ -2350,6 +2351,7 @@ static int patch_alc882(struct hda_codec *codec) switch (codec->vendor_id) { case 0x10ec0882: case 0x10ec0885: + case 0x10ec0900: break; default: /* ALC883 and variants */ -- cgit v1.2.3-70-g09d2 From 03be88ee4ab3acceddca43f11f4d01bcd6edcb93 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 3 Sep 2014 15:52:33 +0300 Subject: ASoC: tlv320aic31xx: Fix 24bit samples with I2S format and 12MHz mclk I2S format requires bitclock to have an exact amount of cycles in a frame for audio to work cleanly. With dsp formats that is not so important. Updates aic31xx_setup_pll() to look for a line in aic31xx_divs table that produces the best match for the bitclock and adds lines to aic31xx_divs for 12MHz mclk and 24bit samples. Signed-off-by: Jyri Sarha Tested-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 51 ++++++++++++++++++++++++++++++---------- 1 file changed, 39 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 0f64c7890eed..aea9e1ff9126 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -189,46 +189,57 @@ static const struct aic31xx_rate_divs aic31xx_divs[] = { /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ /* 8k rate */ {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, + {12000000, 8000, 1, 8, 1920, 128, 32, 3, 128, 32, 3}, {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, /* 11.025k rate */ {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, + {12000000, 11025, 1, 8, 4672, 128, 24, 3, 128, 24, 3}, {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, /* 16k rate */ {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, + {12000000, 16000, 1, 8, 1920, 128, 16, 3, 128, 16, 3}, {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, /* 22.05k rate */ {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, + {12000000, 22050, 1, 8, 4672, 128, 12, 3, 128, 12, 3}, {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, /* 32k rate */ {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, + {12000000, 32000, 1, 8, 1920, 128, 8, 3, 128, 8, 3}, {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, /* 44.1k rate */ {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, + {12000000, 44100, 1, 8, 4672, 128, 6, 3, 128, 6, 3}, {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, /* 48k rate */ {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, + {12000000, 48000, 1, 7, 6800, 96, 5, 4, 96, 5, 4}, {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, /* 88.2k rate */ {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, + {12000000, 88200, 1, 8, 4672, 64, 6, 3, 64, 6, 3}, {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, /* 96k rate */ {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, + {12000000, 96000, 1, 7, 6800, 48, 5, 4, 48, 5, 4}, {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, /* 176.4k rate */ {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, + {12000000, 176400, 1, 8, 4672, 32, 6, 3, 32, 6, 3}, {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, /* 192k rate */ {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, + {12000000, 192000, 1, 7, 6800, 24, 5, 4, 24, 5, 4}, {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, }; @@ -680,7 +691,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, struct snd_pcm_hw_params *params) { struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_score = snd_soc_params_to_frame_size(params); int bclk_n = 0; + int match = -1; int i; /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ @@ -691,15 +704,37 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { if (aic31xx_divs[i].rate == params_rate(params) && - aic31xx_divs[i].mclk == aic31xx->sysclk) - break; + aic31xx_divs[i].mclk == aic31xx->sysclk) { + int s = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) % + snd_soc_params_to_frame_size(params); + int bn = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) / + snd_soc_params_to_frame_size(params); + if (s < bclk_score && bn > 0) { + match = i; + bclk_n = bn; + bclk_score = s; + } + } } - if (i == ARRAY_SIZE(aic31xx_divs)) { - dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + if (match == -1) { + dev_err(codec->dev, + "%s: Sample rate (%u) and format not supported\n", __func__, params_rate(params)); + /* See bellow for details how fix this. */ return -EINVAL; } + if (bclk_score != 0) { + dev_warn(codec->dev, "Can not produce exact bitclock"); + /* This is fine if using dsp format, but if using i2s + there may be trouble. To fix the issue edit the + aic31xx_divs table for your mclk and sample + rate. Details can be found from: + http://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf + Section: 5.6 CLOCK Generation and PLL + */ + } + i = match; /* PLL configuration */ snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, @@ -729,14 +764,6 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); /* Bit clock divider configuration. */ - bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) - / snd_soc_params_to_frame_size(params); - if (bclk_n == 0) { - dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", - __func__); - return -EINVAL; - } - snd_soc_update_bits(codec, AIC31XX_BCLKN, AIC31XX_PLL_MASK, bclk_n); -- cgit v1.2.3-70-g09d2 From fe0a29e163a5d045c73faab682a8dac71c2f8012 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 4 Sep 2014 10:52:53 +0300 Subject: ASoC: davinci-mcasp: Correct rx format unit configuration In case of capture we should not use rotation. The reverse and mask is enough to get the data align correctly from the bus to MCU: Format data from bus after reverse (XRBUF) S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| With this patch all supported formats will work for playback and capture. Reported-by: Jyri Sarha (broken S24_3LE capture) Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/davinci/davinci-mcasp.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c28508da34cf..0062601a63c2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -459,8 +459,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() -- cgit v1.2.3-70-g09d2 From 9a302c32f363e420b6aa6e42c0cd686a495771f6 Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Fri, 5 Sep 2014 16:47:04 +0530 Subject: ASoC: dwc: Update email id of the author I moved from ST Microelectronics and the email-id no longer exists. Update email-id to personal one, Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/dwc/designware_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 25c31f1655f6..a97d27f4d19c 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -4,7 +4,7 @@ * sound/soc/dwc/designware_i2s.c * * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar + * Rajeev Kumar * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any -- cgit v1.2.3-70-g09d2 From b794dbcd31c8b19118b7d02a39453576aa8e78fb Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Tue, 9 Sep 2014 12:27:19 +0530 Subject: ASoC: Update email id of the author I moved from ST Microelectronics and so updating email-id to personal one. Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/codecs/sta529.c | 4 ++-- sound/soc/dwc/designware_i2s.c | 2 +- sound/soc/spear/spear_pcm.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 9aa1323fb2ab..89c748dd3d6e 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -4,7 +4,7 @@ * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver * * Copyright (C) 2012 ST Microelectronics - * Rajeev Kumar + * Rajeev Kumar * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -426,5 +426,5 @@ static struct i2c_driver sta529_i2c_driver = { module_i2c_driver(sta529_i2c_driver); MODULE_DESCRIPTION("ASoC STA529 codec driver"); -MODULE_AUTHOR("Rajeev Kumar "); +MODULE_AUTHOR("Rajeev Kumar "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a97d27f4d19c..e961388e6e9c 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -455,7 +455,7 @@ static struct platform_driver dw_i2s_driver = { module_platform_driver(dw_i2s_driver); -MODULE_AUTHOR("Rajeev Kumar "); +MODULE_AUTHOR("Rajeev Kumar "); MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:designware_i2s"); diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 0e5a8f35d0ad..a7dc3c56f44d 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -4,7 +4,7 @@ * sound/soc/spear/spear_pcm.c * * Copyright (C) 2012 ST Microelectronics - * Rajeev Kumar + * Rajeev Kumar * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -50,6 +50,6 @@ int devm_spear_pcm_platform_register(struct device *dev, } EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register); -MODULE_AUTHOR("Rajeev Kumar "); +MODULE_AUTHOR("Rajeev Kumar "); MODULE_DESCRIPTION("SPEAr PCM DMA module"); MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From 133c2681c4a0c1b589d138c2fdd0f131bdce20ed Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 9 Sep 2014 16:51:49 +0100 Subject: ASoC: samsung-i2s: Check secondary DAI exists before referencing In a couple of places the driver is missing a check to ensure there is a secondary DAI before it de-references the pointer to it, causing a null pointer de-reference. This patch adds a check to avoid this. Signed-off-by: Charles Keepax Acked-by: Sylwester Nawrocki Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/samsung/i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 03eec22f0f46..9d513473b300 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -462,7 +462,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) rfs = 0; - if ((rfs && other->rfs && (other->rfs != rfs)) || + if ((rfs && other && other->rfs && (other->rfs != rfs)) || (any_active(i2s) && (((dir == SND_SOC_CLOCK_IN) && !(mod & MOD_CDCLKCON)) || @@ -762,7 +762,8 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, } else { u32 mod = readl(i2s->addr + I2SMOD); i2s->cdclk_out = !(mod & MOD_CDCLKCON); - other->cdclk_out = i2s->cdclk_out; + if (other) + other->cdclk_out = i2s->cdclk_out; } /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; -- cgit v1.2.3-70-g09d2 From 8f70e515a8bb6a1908b40b786cb43f6491e8da04 Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Wed, 10 Sep 2014 17:54:07 +0800 Subject: ASoC: soc-pcm: fix dpcm_path_get error handling dpcm_path_get may return -ENOMEM when allocating memory for list fails. We should not keep processing path or start up dpcm dai in this case. Signed-off-by: Qiao Zhou Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 6 +++++- sound/soc/soc-pcm.c | 6 +++++- 2 files changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 27c06acce205..3092b58fede6 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -101,7 +101,11 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].runtime = fe_substream->runtime; - if (dpcm_path_get(fe, stream, &list) <= 0) { + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) { + mutex_unlock(&fe->card->mutex); + goto fe_err; + } else if (ret == 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 731fdb5b5f9b..642c86240752 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2352,7 +2352,11 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); fe->dpcm[stream].runtime = fe_substream->runtime; - if (dpcm_path_get(fe, stream, &list) <= 0) { + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) { + mutex_unlock(&fe->card->mutex); + return ret; + } else if (ret == 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } -- cgit v1.2.3-70-g09d2 From 7a9744cb455e6faa287e148394b4b422a6f3c5c4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Sep 2014 12:59:21 +0200 Subject: ALSA: hda - Fix invalid pin powermap without jack detection When a driver is set up without the jack detection explicitly (either by passing a model option or via a specific fixup), the pin powermap of IDT/STAC codecs is set up wrongly, resulting in the silence output. It's because of a logic failure in stac_init_power_map(). It tries to avoid creating a callback for the pins that have other auto-hp and auto-mic callbacks, but the check is done in a wrong way at a wrong time. The stac_init_power_map() should be called after creating other jack detection ctls, and the jack callback should be created only for jack-detectable widgets. This patch fixes the check in stac_init_power_map() and its callee at the right place, after snd_hda_gen_build_controls(). Reported-by: Adam Richter Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea823e1100da..98cd1908c039 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -566,8 +566,8 @@ static void stac_init_power_map(struct hda_codec *codec) if (snd_hda_jack_tbl_get(codec, nid)) continue; if (def_conf == AC_JACK_PORT_COMPLEX && - !(spec->vref_mute_led_nid == nid || - is_jack_detectable(codec, nid))) { + spec->vref_mute_led_nid != nid && + is_jack_detectable(codec, nid)) { snd_hda_jack_detect_enable_callback(codec, nid, STAC_PWR_EVENT, jack_update_power); @@ -4276,11 +4276,18 @@ static int stac_parse_auto_config(struct hda_codec *codec) return err; } - stac_init_power_map(codec); - return 0; } +static int stac_build_controls(struct hda_codec *codec) +{ + int err = snd_hda_gen_build_controls(codec); + + if (err < 0) + return err; + stac_init_power_map(codec); + return 0; +} static int stac_init(struct hda_codec *codec) { @@ -4392,7 +4399,7 @@ static int stac_suspend(struct hda_codec *codec) #endif /* CONFIG_PM */ static const struct hda_codec_ops stac_patch_ops = { - .build_controls = snd_hda_gen_build_controls, + .build_controls = stac_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = stac_init, .free = stac_free, -- cgit v1.2.3-70-g09d2 From 774418253e0ec226ad220c6237bba80fd3f4fbc0 Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Thu, 11 Sep 2014 09:52:46 -0500 Subject: ASoC: cs4265: Fix register address to set the proper data type. The SPDIF control register must be written to set the data type in hw_params not the ADC control register. Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs4265.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index a20b30ca52c0..367242a2757b 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -458,12 +458,12 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (1 << 5)); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 7)); } else { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (3 << 5)); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 7)); } break; @@ -472,7 +472,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, CS4265_DAC_CTL_DIF, 0); snd_soc_update_bits(codec, CS4265_ADC_CTL, CS4265_ADC_DIF, 0); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 6)); break; -- cgit v1.2.3-70-g09d2 From 07833d88314c496f8a136c6e4b4729c69e65b878 Mon Sep 17 00:00:00 2001 From: Jianqun Date: Sat, 13 Sep 2014 08:41:03 +0800 Subject: ASoC: rockchip-i2s: fix master mode set bit error Fix error format set to I2S master or slave mode. Test on RK3288 board with max98090. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 8d8e4b59049f..870a6645c782 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -165,13 +165,14 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; - mask = I2S_CKR_MSS_SLAVE; + mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = I2S_CKR_MSS_SLAVE; + /* Set source clock in Master mode */ + val = I2S_CKR_MSS_MASTER; break; case SND_SOC_DAIFMT_CBM_CFM: - val = I2S_CKR_MSS_MASTER; + val = I2S_CKR_MSS_SLAVE; break; default: return -EINVAL; -- cgit v1.2.3-70-g09d2 From 2f1e93f81cebfa99b668f27cdb14992ff23480a4 Mon Sep 17 00:00:00 2001 From: Jianqun Date: Sat, 13 Sep 2014 08:42:12 +0800 Subject: ASoC: rockchip-i2s: fix registers' property of rockchip i2s controller Reference rockchip I2S controller TRM, modify some registers' property I2S_FIFOLR: read / write, but not volatile, not precious I2S_INTSR: read / write I2S_CLR: volatile, register value will be cleared by read Test on RK3288 with max98090. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 870a6645c782..fb9e05c9f471 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -362,6 +362,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) case I2S_XFER: case I2S_CLR: case I2S_RXDR: + case I2S_FIFOLR: + case I2S_INTSR: return true; default: return false; @@ -371,8 +373,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { - case I2S_FIFOLR: case I2S_INTSR: + case I2S_CLR: return true; default: return false; @@ -382,8 +384,6 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) { switch (reg) { - case I2S_FIFOLR: - return true; default: return false; } -- cgit v1.2.3-70-g09d2 From 85151461f114f2fca386bb8ae6de185461d35d87 Mon Sep 17 00:00:00 2001 From: Michael Trimarchi Date: Thu, 18 Sep 2014 20:38:09 +0200 Subject: ASoC: fsl_ssi: fix kernel panic in probe function code can raise a panic when the ssi_private->pdev is null [...] /* * If codec-handle property is missing from SSI node, we assume * that the machine driver uses new binding which does not require * SSI driver to trigger machine driver's probe. */ if (!of_get_property(np, "codec-handle", NULL)) goto done; [...] ssi_private->pdev = platform_device_register_data(&pdev->dev, name, 0, NULL, 0); [...] done: if (ssi_private->dai_fmt) _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt); Proposal was to not use ssi_private->pdev->dev here but adding a new parameter of *dev pointer to this _set_dai_fmt() -- passing pdev->dev in probe() and cpu_dai->dev in fsl_ssi_set_dai_fmt(). Signed-off-by: Michael Trimarchi Reported-by: Jean-Michel Hautbois Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb5776a39b..de6ab06f58a5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -748,8 +748,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, return 0; } -static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, - unsigned int fmt) +static int _fsl_ssi_set_dai_fmt(struct device *dev, + struct fsl_ssi_private *ssi_private, + unsigned int fmt) { struct regmap *regs = ssi_private->regs; u32 strcr = 0, stcr, srcr, scr, mask; @@ -758,7 +759,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt; if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) { - dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n"); + dev_err(dev, "baudclk is missing which is necessary for master mode\n"); return -EINVAL; } @@ -913,7 +914,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - return _fsl_ssi_set_dai_fmt(ssi_private, fmt); + return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt); } /** @@ -1387,7 +1388,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) done: if (ssi_private->dai_fmt) - _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt); + _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, + ssi_private->dai_fmt); return 0; -- cgit v1.2.3-70-g09d2 From 2e4ec1c0b8791f3556576a19cf56941c3d2a90fc Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Fri, 12 Sep 2014 09:41:31 +0800 Subject: ASoC: soc-compress: fix double unlock of fe card mutex Fix double unlock of fe card mutex introduced by patch 8f70e515a8bb "ASoC: soc-pcm: fix dpcm_path_get error handling" The first unlock is at line 106, and the unlock is at line 149. we should remove the first unlock. Reported-by: Dan Carpenter Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3092b58fede6..cecfab3cc948 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -102,13 +102,11 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].runtime = fe_substream->runtime; ret = dpcm_path_get(fe, stream, &list); - if (ret < 0) { - mutex_unlock(&fe->card->mutex); + if (ret < 0) goto fe_err; - } else if (ret == 0) { + else if (ret == 0) dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); - } /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); -- cgit v1.2.3-70-g09d2 From b7a297677540789b8fb35a6ce66c500739fb4bf9 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 26 Sep 2014 11:06:39 +0800 Subject: ASoC: rt286: Correct default value This patch corrects some incorrect default value in the cache. Signed-off-by: Bard Liao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt286.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e4f6102efc1a..7a6608404d04 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -51,7 +51,7 @@ static struct reg_default rt286_index_def[] = { { 0x04, 0xaf01 }, { 0x08, 0x000d }, { 0x09, 0xd810 }, - { 0x0a, 0x0060 }, + { 0x0a, 0x0120 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, @@ -60,7 +60,7 @@ static struct reg_default rt286_index_def[] = { { 0x33, 0x0208 }, { 0x49, 0x0004 }, { 0x4f, 0x50e9 }, - { 0x50, 0x2c00 }, + { 0x50, 0x2000 }, { 0x63, 0x2902 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, @@ -104,7 +104,6 @@ static const struct reg_default rt286_reg[] = { { 0x02170700, 0x00000000 }, { 0x02270100, 0x00000000 }, { 0x02370100, 0x00000000 }, - { 0x02040000, 0x00004002 }, { 0x01870700, 0x00000020 }, { 0x00830000, 0x000000c3 }, { 0x00930000, 0x000000c3 }, -- cgit v1.2.3-70-g09d2 From 66d627d554a4284dad00b2039efd18e1c129cc2f Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 26 Sep 2014 11:06:40 +0800 Subject: ASoC: rt286: Fix sync function We try to write index registers into cache when we write an index register, but we change the reg value before updating the cache. As a result, the cache is never be updated. This patch will fix this issue. Signed-off-by: Bard Liao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 7a6608404d04..b86b426f159d 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -191,7 +191,6 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) /*handle index registers*/ if (reg <= 0xff) { rt286_hw_write(client, RT286_COEF_INDEX, reg); - reg = RT286_PROC_COEF; for (i = 0; i < INDEX_CACHE_SIZE; i++) { if (reg == rt286->index_cache[i].reg) { rt286->index_cache[i].def = value; @@ -199,6 +198,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) } } + reg = RT286_PROC_COEF; } data[0] = (reg >> 24) & 0xff; -- cgit v1.2.3-70-g09d2 From 6596aa047b624aeec2ea321962cfdecf9953a383 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Sun, 28 Sep 2014 17:29:37 +0800 Subject: ASoC: core: fix possible ZERO_SIZE_PTR pointer dereferencing error. Since we cannot make sure the 'params->num_regs' will always be none zero here, and then if it equals to zero, the kmemdup() will return ZERO_SIZE_PTR, which equals to ((void *)16). So this patch fix this with just doing the zero check before calling kmemdup(). Signed-off-by: Xiubo Li Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d4bfd4a9076f..ae48f1013e80 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3203,7 +3203,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, unsigned int val, mask; void *data; - if (!component->regmap) + if (!component->regmap || !params->num_regs) return -EINVAL; len = params->num_regs * component->val_bytes; -- cgit v1.2.3-70-g09d2