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authorLinus Torvalds <torvalds@linux-foundation.org>2011-10-28 14:25:01 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2011-10-28 14:25:01 -0700
commit68d99b2c8efcb6ed3807a55569300c53b5f88be5 (patch)
treef189c8f2132d3668a2f0e503f5c3f8695b26a1c8 /sound/soc/au1x/i2sc.c
parent0e59e7e7feb5a12938fbf9135147eeda3238c6c4 (diff)
parent8128c9f21509f9a8b6da94ac432d845dda458406 (diff)
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits) ALSA: hda - Fix ADC input-amp handling for Cx20549 codec ALSA: hda - Keep EAPD turned on for old Conexant chips ALSA: hda/realtek - Fix missing volume controls with ALC260 ASoC: wm8940: Properly set codec->dapm.bias_level ALSA: hda - Fix pin-config for ASUS W90V ALSA: hda - Fix surround/CLFE headphone and speaker pins order ALSA: hda - Fix typo ALSA: Update the sound git tree URL ALSA: HDA: Add new revision for ALC662 ASoC: max98095: Convert codec->hw_write to snd_soc_write ASoC: keep pointer to resource so it can be freed ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2 ASoC: da7210: Add support for line out and DAC ASoC: da7210: Add support for DAPM ALSA: hda/realtek - Fix DAC assignments of multiple speakers ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value ASoC: Set sgtl5000->ldo in ldo_regulator_register ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture ...
Diffstat (limited to 'sound/soc/au1x/i2sc.c')
-rw-r--r--sound/soc/au1x/i2sc.c349
1 files changed, 349 insertions, 0 deletions
diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c
new file mode 100644
index 000000000000..6bcf48f5884c
--- /dev/null
+++ b/sound/soc/au1x/i2sc.c
@@ -0,0 +1,349 @@
+/*
+ * Au1000/Au1500/Au1100 I2S controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * Note: clock supplied to the I2S controller must be 256x samplerate.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+
+#include "psc.h"
+
+#define I2S_RXTX 0x00
+#define I2S_CFG 0x04
+#define I2S_ENABLE 0x08
+
+#define CFG_XU (1 << 25) /* tx underflow */
+#define CFG_XO (1 << 24)
+#define CFG_RU (1 << 23)
+#define CFG_RO (1 << 22)
+#define CFG_TR (1 << 21)
+#define CFG_TE (1 << 20)
+#define CFG_TF (1 << 19)
+#define CFG_RR (1 << 18)
+#define CFG_RF (1 << 17)
+#define CFG_ICK (1 << 12) /* clock invert */
+#define CFG_PD (1 << 11) /* set to make I2SDIO INPUT */
+#define CFG_LB (1 << 10) /* loopback */
+#define CFG_IC (1 << 9) /* word select invert */
+#define CFG_FM_I2S (0 << 7) /* I2S format */
+#define CFG_FM_LJ (1 << 7) /* left-justified */
+#define CFG_FM_RJ (2 << 7) /* right-justified */
+#define CFG_FM_MASK (3 << 7)
+#define CFG_TN (1 << 6) /* tx fifo en */
+#define CFG_RN (1 << 5) /* rx fifo en */
+#define CFG_SZ_8 (0x08)
+#define CFG_SZ_16 (0x10)
+#define CFG_SZ_18 (0x12)
+#define CFG_SZ_20 (0x14)
+#define CFG_SZ_24 (0x18)
+#define CFG_SZ_MASK (0x1f)
+#define EN_D (1 << 1) /* DISable */
+#define EN_CE (1 << 0) /* clock enable */
+
+/* only limited by clock generator and board design */
+#define AU1XI2SC_RATES \
+ SNDRV_PCM_RATE_CONTINUOUS
+
+#define AU1XI2SC_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \
+ SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_U18_3BE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3BE | SNDRV_PCM_FMTBIT_U20_3BE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \
+ SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE | \
+ 0)
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+ return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+ __raw_writel(v, ctx->mmio + reg);
+ wmb();
+}
+
+static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(cpu_dai);
+ unsigned long c;
+ int ret;
+
+ ret = -EINVAL;
+ c = ctx->cfg;
+
+ c &= ~CFG_FM_MASK;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ c |= CFG_FM_I2S;
+ break;
+ case SND_SOC_DAIFMT_MSB:
+ c |= CFG_FM_RJ;
+ break;
+ case SND_SOC_DAIFMT_LSB:
+ c |= CFG_FM_LJ;
+ break;
+ default:
+ goto out;
+ }
+
+ c &= ~(CFG_IC | CFG_ICK); /* IB-IF */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ c |= CFG_IC | CFG_ICK;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ c |= CFG_IC;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ c |= CFG_ICK;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ break;
+ default:
+ goto out;
+ }
+
+ /* I2S controller only supports master */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */
+ break;
+ default:
+ goto out;
+ }
+
+ ret = 0;
+ ctx->cfg = c;
+out:
+ return ret;
+}
+
+static int au1xi2s_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ int stype = SUBSTREAM_TYPE(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ /* power up */
+ WR(ctx, I2S_ENABLE, EN_D | EN_CE);
+ WR(ctx, I2S_ENABLE, EN_CE);
+ ctx->cfg |= (stype == PCM_TX) ? CFG_TN : CFG_RN;
+ WR(ctx, I2S_CFG, ctx->cfg);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ ctx->cfg &= ~((stype == PCM_TX) ? CFG_TN : CFG_RN);
+ WR(ctx, I2S_CFG, ctx->cfg);
+ WR(ctx, I2S_ENABLE, EN_D); /* power off */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static unsigned long msbits_to_reg(int msbits)
+{
+ switch (msbits) {
+ case 8:
+ return CFG_SZ_8;
+ case 16:
+ return CFG_SZ_16;
+ case 18:
+ return CFG_SZ_18;
+ case 20:
+ return CFG_SZ_20;
+ case 24:
+ return CFG_SZ_24;
+ }
+ return 0;
+}
+
+static int au1xi2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ unsigned long v;
+
+ v = msbits_to_reg(params->msbits);
+ if (!v)
+ return -EINVAL;
+
+ ctx->cfg &= ~CFG_SZ_MASK;
+ ctx->cfg |= v;
+ return 0;
+}
+
+static int au1xi2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+ return 0;
+}
+
+static const struct snd_soc_dai_ops au1xi2s_dai_ops = {
+ .startup = au1xi2s_startup,
+ .trigger = au1xi2s_trigger,
+ .hw_params = au1xi2s_hw_params,
+ .set_fmt = au1xi2s_set_fmt,
+};
+
+static struct snd_soc_dai_driver au1xi2s_dai_driver = {
+ .symmetric_rates = 1,
+ .playback = {
+ .rates = AU1XI2SC_RATES,
+ .formats = AU1XI2SC_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AU1XI2SC_RATES,
+ .formats = AU1XI2SC_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = &au1xi2s_dai_ops,
+};
+
+static int __devinit au1xi2s_drvprobe(struct platform_device *pdev)
+{
+ int ret;
+ struct resource *iores, *dmares;
+ struct au1xpsc_audio_data *ctx;
+
+ ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!iores) {
+ ret = -ENODEV;
+ goto out0;
+ }
+
+ ret = -EBUSY;
+ if (!request_mem_region(iores->start, resource_size(iores),
+ pdev->name))
+ goto out0;
+
+ ctx->mmio = ioremap_nocache(iores->start, resource_size(iores));
+ if (!ctx->mmio)
+ goto out1;
+
+ dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!dmares)
+ goto out2;
+ ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start;
+
+ dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!dmares)
+ goto out2;
+ ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start;
+
+ platform_set_drvdata(pdev, ctx);
+
+ ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver);
+ if (ret)
+ goto out2;
+
+ return 0;
+
+out2:
+ iounmap(ctx->mmio);
+out1:
+ release_mem_region(iores->start, resource_size(iores));
+out0:
+ kfree(ctx);
+ return ret;
+}
+
+static int __devexit au1xi2s_drvremove(struct platform_device *pdev)
+{
+ struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+ struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */
+
+ iounmap(ctx->mmio);
+ release_mem_region(r->start, resource_size(r));
+ kfree(ctx);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xi2s_drvsuspend(struct device *dev)
+{
+ struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+ WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */
+
+ return 0;
+}
+
+static int au1xi2s_drvresume(struct device *dev)
+{
+ return 0;
+}
+
+static const struct dev_pm_ops au1xi2sc_pmops = {
+ .suspend = au1xi2s_drvsuspend,
+ .resume = au1xi2s_drvresume,
+};
+
+#define AU1XI2SC_PMOPS (&au1xi2sc_pmops)
+
+#else
+
+#define AU1XI2SC_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xi2s_driver = {
+ .driver = {
+ .name = "alchemy-i2sc",
+ .owner = THIS_MODULE,
+ .pm = AU1XI2SC_PMOPS,
+ },
+ .probe = au1xi2s_drvprobe,
+ .remove = __devexit_p(au1xi2s_drvremove),
+};
+
+static int __init au1xi2s_load(void)
+{
+ return platform_driver_register(&au1xi2s_driver);
+}
+
+static void __exit au1xi2s_unload(void)
+{
+ platform_driver_unregister(&au1xi2s_driver);
+}
+
+module_init(au1xi2s_load);
+module_exit(au1xi2s_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 I2S ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");