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-rw-r--r--sound/soc/codecs/Makefile1
-rw-r--r--sound/soc/codecs/ak4535.c3
-rw-r--r--sound/soc/codecs/twl4030.c61
-rw-r--r--sound/soc/codecs/twl4030.h1
-rw-r--r--sound/soc/codecs/wm8580.c16
-rw-r--r--sound/soc/codecs/wm9705.c37
-rw-r--r--sound/soc/fsl/fsl_dma.c29
-rw-r--r--sound/soc/fsl/fsl_ssi.c99
-rw-r--r--sound/soc/omap/n810.c4
-rw-r--r--sound/soc/omap/omap-mcbsp.c23
-rw-r--r--sound/soc/omap/omap-mcbsp.h3
-rw-r--r--sound/soc/omap/omap-pcm.c5
-rw-r--r--sound/soc/omap/omap-pcm.h3
-rw-r--r--sound/soc/omap/osk5912.c4
-rw-r--r--sound/soc/pxa/Kconfig10
-rw-r--r--sound/soc/pxa/Makefile2
-rw-r--r--sound/soc/pxa/magician.c558
-rw-r--r--sound/soc/pxa/palm27x.c27
-rw-r--r--sound/soc/pxa/pxa-ssp.c49
-rw-r--r--sound/soc/s3c24xx/Kconfig6
-rw-r--r--sound/soc/s3c24xx/jive_wm8750.c12
-rw-r--r--sound/soc/s3c24xx/s3c-i2s-v2.c21
-rw-r--r--sound/soc/s3c24xx/s3c2412-i2s.c2
-rw-r--r--sound/soc/soc-core.c20
24 files changed, 880 insertions, 116 deletions
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 030d2454725f..f2653803ede8 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -56,7 +56,6 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
-obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o
obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 1f63d387a2f4..dd3380202766 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -659,7 +659,8 @@ static int ak4535_remove(struct platform_device *pdev)
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- i2c_unregister_device(codec->control_data);
+ if (codec->control_data)
+ i2c_unregister_device(codec->control_data);
i2c_del_driver(&ak4535_i2c_driver);
#endif
kfree(codec->private_data);
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 97738e2ece04..921b205de28a 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -122,6 +122,9 @@ struct twl4030_priv {
unsigned int bypass_state;
unsigned int codec_powered;
unsigned int codec_muted;
+
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
};
/*
@@ -1217,6 +1220,52 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
+static int twl4030_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ struct twl4030_priv *twl4030 = codec->private_data;
+
+ /* If we already have a playback or capture going then constrain
+ * this substream to match it.
+ */
+ if (twl4030->master_substream) {
+ struct snd_pcm_runtime *master_runtime;
+ master_runtime = twl4030->master_substream->runtime;
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
+
+ twl4030->slave_substream = substream;
+ } else
+ twl4030->master_substream = substream;
+
+ return 0;
+}
+
+static void twl4030_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ struct twl4030_priv *twl4030 = codec->private_data;
+
+ if (twl4030->master_substream == substream)
+ twl4030->master_substream = twl4030->slave_substream;
+
+ twl4030->slave_substream = NULL;
+}
+
static int twl4030_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -1224,8 +1273,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
+ struct twl4030_priv *twl4030 = codec->private_data;
u8 mode, old_mode, format, old_format;
+ if (substream == twl4030->slave_substream)
+ /* Ignoring hw_params for slave substream */
+ return 0;
+
/* bit rate */
old_mode = twl4030_read_reg_cache(codec,
TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
@@ -1259,6 +1313,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
case 48000:
mode |= TWL4030_APLL_RATE_48000;
break;
+ case 96000:
+ mode |= TWL4030_APLL_RATE_96000;
+ break;
default:
printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
params_rate(params));
@@ -1384,6 +1441,8 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
static struct snd_soc_dai_ops twl4030_dai_ops = {
+ .startup = twl4030_startup,
+ .shutdown = twl4030_shutdown,
.hw_params = twl4030_hw_params,
.set_sysclk = twl4030_set_dai_sysclk,
.set_fmt = twl4030_set_dai_fmt,
@@ -1395,7 +1454,7 @@ struct snd_soc_dai twl4030_dai = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = TWL4030_RATES,
+ .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
.formats = TWL4030_FORMATS,},
.capture = {
.stream_name = "Capture",
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index 33dbb144dad1..cb63765db1df 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -109,6 +109,7 @@
#define TWL4030_APLL_RATE_32000 0x80
#define TWL4030_APLL_RATE_44100 0x90
#define TWL4030_APLL_RATE_48000 0xA0
+#define TWL4030_APLL_RATE_96000 0xE0
#define TWL4030_SEL_16K 0x04
#define TWL4030_CODECPDZ 0x02
#define TWL4030_OPT_MODE 0x01
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 442ea6f160fc..9f6be3d31ac0 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- int reg = kcontrol->private_value & 0xff;
- int reg2 = (kcontrol->private_value >> 24) & 0xff;
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
int ret;
u16 val;
@@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
return 0;
}
-#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
+#define SOC_WM8580_OUT_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 |\
SNDRV_CTL_ELEM_ACCESS_READWRITE, \
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw_2r, \
.get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
- .private_value = (reg_left) | ((shift) << 8) | \
- ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+ .max = xmax, .invert = xinvert} }
static const struct snd_kcontrol_new wm8580_snd_controls[] = {
SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
@@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
reg = wm8580_read(codec, WM8580_PLLA4 + offset);
reg &= ~0x3f;
reg |= pll_div.prescale | pll_div.postscale << 1 |
- pll_div.freqmode << 4;
+ pll_div.freqmode << 3;
wm8580_write(codec, WM8580_PLLA4 + offset, reg);
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 3265817c5c26..c2d1a7a18fa3 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -317,6 +317,41 @@ static int wm9705_reset(struct snd_soc_codec *codec)
return -EIO;
}
+#ifdef CONFIG_PM
+static int wm9705_soc_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff);
+
+ return 0;
+}
+
+static int wm9705_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i, ret;
+ u16 *cache = codec->reg_cache;
+
+ ret = wm9705_reset(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "could not reset AC97 codec\n");
+ return ret;
+ }
+
+ for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) {
+ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
+ }
+
+ return 0;
+}
+#else
+#define wm9705_soc_suspend NULL
+#define wm9705_soc_resume NULL
+#endif
+
static int wm9705_soc_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -407,6 +442,8 @@ static int wm9705_soc_remove(struct platform_device *pdev)
struct snd_soc_codec_device soc_codec_dev_wm9705 = {
.probe = wm9705_soc_probe,
.remove = wm9705_soc_remove,
+ .suspend = wm9705_soc_suspend,
+ .resume = wm9705_soc_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705);
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index b3eb8570cd7b..b1a3a278819f 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -300,7 +300,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = fsl_dma_dmamask;
- ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev,
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[0].substream->dma_buffer);
if (ret) {
@@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
return -ENOMEM;
}
- ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev,
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[1].substream->dma_buffer);
if (ret) {
@@ -418,7 +418,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
return -EBUSY;
}
- dma_private = dma_alloc_coherent(substream->pcm->dev,
+ dma_private = dma_alloc_coherent(substream->pcm->card->dev,
sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL);
if (!dma_private) {
dev_err(substream->pcm->card->dev,
@@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
dev_err(substream->pcm->card->dev,
"can't register ISR for IRQ %u (ret=%i)\n",
dma_private->irq, ret);
- dma_free_coherent(substream->pcm->dev,
+ dma_free_coherent(substream->pcm->card->dev,
sizeof(struct fsl_dma_private),
dma_private, dma_private->ld_buf_phys);
return ret;
@@ -697,6 +697,23 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream)
else
position = in_be32(&dma_channel->dar);
+ /*
+ * When capture is started, the SSI immediately starts to fill its FIFO.
+ * This means that the DMA controller is not started until the FIFO is
+ * full. However, ALSA calls this function before that happens, when
+ * MR.DAR is still zero. In this case, just return zero to indicate
+ * that nothing has been received yet.
+ */
+ if (!position)
+ return 0;
+
+ if ((position < dma_private->dma_buf_phys) ||
+ (position > dma_private->dma_buf_end)) {
+ dev_err(substream->pcm->card->dev,
+ "dma pointer is out of range, halting stream\n");
+ return SNDRV_PCM_POS_XRUN;
+ }
+
frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys);
/*
@@ -761,13 +778,13 @@ static int fsl_dma_close(struct snd_pcm_substream *substream)
free_irq(dma_private->irq, dma_private);
if (dma_private->ld_buf_phys) {
- dma_unmap_single(substream->pcm->dev,
+ dma_unmap_single(substream->pcm->card->dev,
dma_private->ld_buf_phys,
sizeof(dma_private->link), DMA_TO_DEVICE);
}
/* Deallocate the fsl_dma_private structure */
- dma_free_coherent(substream->pcm->dev,
+ dma_free_coherent(substream->pcm->card->dev,
sizeof(struct fsl_dma_private),
dma_private, dma_private->ld_buf_phys);
substream->runtime->private_data = NULL;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 169bca295b78..3711d8454d96 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -60,6 +60,13 @@
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE)
#endif
+/* SIER bitflag of interrupts to enable */
+#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \
+ CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \
+ CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \
+ CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \
+ CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN)
+
/**
* fsl_ssi_private: per-SSI private data
*
@@ -140,7 +147,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
were interrupted for. We mask it with the Interrupt Enable register
so that we only check for events that we're interested in.
*/
- sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier);
+ sisr = in_be32(&ssi->sisr) & SIER_FLAGS;
if (sisr & CCSR_SSI_SISR_RFRC) {
ssi_private->stats.rfrc++;
@@ -324,12 +331,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
*/
/* 4. Enable the interrupts and DMA requests */
- out_be32(&ssi->sier,
- CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE |
- CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN |
- CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN |
- CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE |
- CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN);
+ out_be32(&ssi->sier, SIER_FLAGS);
/*
* Set the watermark for transmit FIFI 0 and receive FIFO 0. We
@@ -466,28 +468,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
case SNDRV_PCM_TRIGGER_START:
clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
setbits32(&ssi->scr,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
- } else {
- long timeout = jiffies + 10;
-
+ else
setbits32(&ssi->scr,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
-
- /* Wait until the SSI has filled its FIFO. Without this
- * delay, ALSA complains about overruns. When the FIFO
- * is full, the DMA controller initiates its first
- * transfer. Until then, however, the DMA's DAR
- * register is zero, which translates to an
- * out-of-bounds pointer. This makes ALSA think an
- * overrun has occurred.
- */
- while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) &&
- (jiffies < timeout));
- if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0))
- return -EIO;
- }
break;
case SNDRV_PCM_TRIGGER_STOP:
@@ -606,39 +592,52 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
.ops = &fsl_ssi_dai_ops,
};
+/* 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.
+ */
+#define SIER_SHOW(flag, name) \
+ do { \
+ if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \
+ length += sprintf(buf + length, #name "=%u\n", \
+ ssi_private->stats.name); \
+ } while (0)
+
+
/**
* fsl_sysfs_ssi_show: display SSI statistics
*
- * Display the statistics for the current SSI device.
+ * Display the statistics for the current SSI device. To avoid confusion,
+ * we only show those counts that are enabled.
*/
static ssize_t fsl_sysfs_ssi_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct fsl_ssi_private *ssi_private =
- container_of(attr, struct fsl_ssi_private, dev_attr);
- ssize_t length;
-
- length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc);
- length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc);
- length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau);
- length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu);
- length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt);
- length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1);
- length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0);
- length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1);
- length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0);
- length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1);
- length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0);
- length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1);
- length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0);
- length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs);
- length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs);
- length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls);
- length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls);
- length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1);
- length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0);
- length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1);
- length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0);
+ container_of(attr, struct fsl_ssi_private, dev_attr);
+ ssize_t length = 0;
+
+ SIER_SHOW(RFRC_EN, rfrc);
+ SIER_SHOW(TFRC_EN, tfrc);
+ SIER_SHOW(CMDAU_EN, cmdau);
+ SIER_SHOW(CMDDU_EN, cmddu);
+ SIER_SHOW(RXT_EN, rxt);
+ SIER_SHOW(RDR1_EN, rdr1);
+ SIER_SHOW(RDR0_EN, rdr0);
+ SIER_SHOW(TDE1_EN, tde1);
+ SIER_SHOW(TDE0_EN, tde0);
+ SIER_SHOW(ROE1_EN, roe1);
+ SIER_SHOW(ROE0_EN, roe0);
+ SIER_SHOW(TUE1_EN, tue1);
+ SIER_SHOW(TUE0_EN, tue0);
+ SIER_SHOW(TFS_EN, tfs);
+ SIER_SHOW(RFS_EN, rfs);
+ SIER_SHOW(TLS_EN, tls);
+ SIER_SHOW(RLS_EN, rls);
+ SIER_SHOW(RFF1_EN, rff1);
+ SIER_SHOW(RFF0_EN, rff0);
+ SIER_SHOW(TFE1_EN, tfe1);
+ SIER_SHOW(TFE0_EN, tfe0);
return length;
}
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index a6d1178ce128..91ef17992de5 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -3,7 +3,7 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@@ -417,6 +417,6 @@ static void __exit n810_soc_exit(void)
module_init(n810_soc_init);
module_exit(n810_soc_exit);
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
MODULE_DESCRIPTION("ALSA SoC Nokia N810");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d6882be33452..912614283848 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -3,7 +3,8 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Peter Ujfalusi <peter.ujfalusi@nokia.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@@ -146,6 +147,17 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
int err = 0;
+ if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) {
+ /*
+ * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
+ * Set constraint for minimum buffer size to the same than FIFO
+ * size in order to avoid underruns in playback startup because
+ * HW is keeping the DMA request active until FIFO is filled.
+ */
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX);
+ }
+
if (!cpu_dai->active)
err = omap_mcbsp_request(mcbsp_data->bus_id);
@@ -272,7 +284,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
break;
case SND_SOC_DAIFMT_DSP_B:
regs->srgr2 |= FPER(wlen * channels - 1);
- regs->srgr1 |= FWID(wlen * channels - 2);
+ regs->srgr1 |= FWID(0);
break;
}
@@ -291,6 +303,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
{
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+ unsigned int temp_fmt = fmt;
if (mcbsp_data->configured)
return 0;
@@ -317,6 +330,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
/* 0-bit data delay */
regs->rcr2 |= RDATDLY(0);
regs->xcr2 |= XDATDLY(0);
+ /* Invert FS polarity configuration */
+ temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
break;
default:
/* Unsupported data format */
@@ -340,7 +355,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
}
/* Set bit clock (CLKX/CLKR) and FS polarities */
- switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
/*
* Normal BCLK + FS.
@@ -518,6 +533,6 @@ static void __exit snd_omap_mcbsp_exit(void)
}
module_exit(snd_omap_mcbsp_exit);
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
MODULE_DESCRIPTION("OMAP I2S SoC Interface");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index df7ad13ba73d..c8147aace813 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -3,7 +3,8 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Peter Ujfalusi <peter.ujfalusi@nokia.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 1bdbb0427183..07cf7f46b584 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -3,7 +3,8 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Peter Ujfalusi <peter.ujfalusi@nokia.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@@ -367,6 +368,6 @@ static void __exit omap_soc_platform_exit(void)
}
module_exit(omap_soc_platform_exit);
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
MODULE_DESCRIPTION("OMAP PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index e4369bdfd77d..8d9d26916b05 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -3,7 +3,8 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Peter Ujfalusi <peter.ujfalusi@nokia.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index a952a4eb3361..a4e149b7f0eb 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -62,7 +62,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
/* Set codec DAI configuration */
err = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_DSP_B |
- SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0) {
printk(KERN_ERR "can't set codec DAI configuration\n");
@@ -72,7 +72,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
/* Set cpu DAI configuration */
err = snd_soc_dai_set_fmt(cpu_dai,
SND_SOC_DAIFMT_DSP_B |
- SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0) {
printk(KERN_ERR "can't set cpu DAI configuration\n");
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 5998ab366e83..ad8a10fe6298 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -116,6 +116,16 @@ config SND_SOC_ZYLONITE
Say Y if you want to add support for SoC audio on the
Marvell Zylonite reference platform.
+config SND_PXA2XX_SOC_MAGICIAN
+ tristate "SoC Audio support for HTC Magician"
+ depends on SND_PXA2XX_SOC && MACH_MAGICIAN
+ select SND_PXA2XX_SOC_I2S
+ select SND_PXA_SOC_SSP
+ select SND_SOC_UDA1380
+ help
+ Say Y if you want to add support for SoC audio on the
+ HTC Magician.
+
config SND_PXA2XX_SOC_MIOA701
tristate "SoC Audio support for MIO A701"
depends on SND_PXA2XX_SOC && MACH_MIOA701
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 8ed881c5e5cc..4b90c3ccae45 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -20,6 +20,7 @@ snd-soc-spitz-objs := spitz.o
snd-soc-em-x270-objs := em-x270.o
snd-soc-palm27x-objs := palm27x.o
snd-soc-zylonite-objs := zylonite.o
+snd-soc-magician-objs := magician.o
snd-soc-mioa701-objs := mioa701_wm9713.o
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
@@ -31,5 +32,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
new file mode 100644
index 000000000000..0625c342a1c9
--- /dev/null
+++ b/sound/soc/pxa/magician.c
@@ -0,0 +1,558 @@
+/*
+ * SoC audio for HTC Magician
+ *
+ * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
+ *
+ * based on spitz.c,
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * 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 <linux/module.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/magician.h>
+#include <asm/mach-types.h>
+#include "../codecs/uda1380.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+#include "pxa-ssp.h"
+
+#define MAGICIAN_MIC 0
+#define MAGICIAN_MIC_EXT 1
+
+static int magician_hp_switch;
+static int magician_spk_switch = 1;
+static int magician_in_sel = MAGICIAN_MIC;
+
+static void magician_ext_control(struct snd_soc_codec *codec)
+{
+ if (magician_spk_switch)
+ snd_soc_dapm_enable_pin(codec, "Speaker");
+ else
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+ if (magician_hp_switch)
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ else
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+ switch (magician_in_sel) {
+ case MAGICIAN_MIC:
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_enable_pin(codec, "Call Mic");
+ break;
+ case MAGICIAN_MIC_EXT:
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ break;
+ }
+
+ snd_soc_dapm_sync(codec);
+}
+
+static int magician_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->card->codec;
+
+ /* check the jack status at stream startup */
+ magician_ext_control(codec);
+
+ return 0;
+}
+
+/*
+ * Magician uses SSP port for playback.
+ */
+static int magician_playback_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->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int acps, acds, width, rate;
+ unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+ int ret = 0;
+
+ rate = params_rate(params);
+ width = snd_pcm_format_physical_width(params_format(params));
+
+ /*
+ * rate = SSPSCLK / (2 * width(16 or 32))
+ * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
+ */
+ switch (params_rate(params)) {
+ case 8000:
+ /* off by a factor of 2: bug in the PXA27x audio clock? */
+ acps = 32842000;
+ switch (width) {
+ case 16:
+ /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_16;
+ break;
+ case 32:
+ /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_8;
+ }
+ break;
+ case 11025:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_4;
+ break;
+ case 32:
+ /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ }
+ break;
+ case 22050:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ case 32:
+ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 44100:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ case 32:
+ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 48000:
+ acps = 12235000;
+ switch (width) {
+ case 16:
+ /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ case 32:
+ /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 96000:
+ acps = 12235000;
+ switch (width) {
+ case 16:
+ /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ break;
+ case 32:
+ /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ div4 = PXA_SSP_CLK_SCDB_1;
+ break;
+ }
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
+ if (ret < 0)
+ return ret;
+
+ /* set audio clock as clock source */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ /* set the SSP audio system clock ACDS divider */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai,
+ PXA_SSP_AUDIO_DIV_ACDS, acds);
+ if (ret < 0)
+ return ret;
+
+ /* set the SSP audio system clock SCDB divider4 */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai,
+ PXA_SSP_AUDIO_DIV_SCDB, div4);
+ if (ret < 0)
+ return ret;
+
+ /* set SSP audio pll clock */
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+/*
+ * Magician uses I2S for capture.
+ */
+static int magician_capture_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->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret = 0;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as output */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops magician_capture_ops = {
+ .startup = magician_startup,
+ .hw_params = magician_capture_hw_params,
+};
+
+static struct snd_soc_ops magician_playback_ops = {
+ .startup = magician_startup,
+ .hw_params = magician_playback_hw_params,
+};
+
+static int magician_get_hp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_hp_switch;
+ return 0;
+}
+
+static int magician_set_hp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (magician_hp_switch == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_hp_switch = ucontrol->value.integer.value[0];
+ magician_ext_control(codec);
+ return 1;
+}
+
+static int magician_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_spk_switch;
+ return 0;
+}
+
+static int magician_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (magician_spk_switch == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_spk_switch = ucontrol->value.integer.value[0];
+ magician_ext_control(codec);
+ return 1;
+}
+
+static int magician_get_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_in_sel;
+ return 0;
+}
+
+static int magician_set_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (magician_in_sel == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_in_sel = ucontrol->value.integer.value[0];
+
+ switch (magician_in_sel) {
+ case MAGICIAN_MIC:
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
+ break;
+ case MAGICIAN_MIC_EXT:
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
+ }
+
+ return 1;
+}
+
+static int magician_spk_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int magician_hp_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int magician_mic_bias(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+/* magician machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
+ SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
+ SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
+ SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
+};
+
+/* magician machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* Headphone connected to VOUTL, VOUTR */
+ {"Headphone Jack", NULL, "VOUTL"},
+ {"Headphone Jack", NULL, "VOUTR"},
+
+ /* Speaker connected to VOUTL, VOUTR */
+ {"Speaker", NULL, "VOUTL"},
+ {"Speaker", NULL, "VOUTR"},
+
+ /* Mics are connected to VINM */
+ {"VINM", NULL, "Headset Mic"},
+ {"VINM", NULL, "Call Mic"},
+};
+
+static const char *input_select[] = {"Call Mic", "Headset Mic"};
+static const struct soc_enum magician_in_sel_enum =
+ SOC_ENUM_SINGLE_EXT(2, input_select);
+
+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
+ SOC_SINGLE_BOOL_EXT("Headphone Switch",
+ (unsigned long)&magician_hp_switch,
+ magician_get_hp, magician_set_hp),
+ SOC_SINGLE_BOOL_EXT("Speaker Switch",
+ (unsigned long)&magician_spk_switch,
+ magician_get_spk, magician_set_spk),
+ SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
+ magician_get_input, magician_set_input),
+};
+
+/*
+ * Logic for a uda1380 as connected on a HTC Magician
+ */
+static int magician_uda1380_init(struct snd_soc_codec *codec)
+{
+ int err;
+
+ /* NC codec pins */
+ snd_soc_dapm_nc_pin(codec, "VOUTLHP");
+ snd_soc_dapm_nc_pin(codec, "VOUTRHP");
+
+ /* FIXME: is anything connected here? */
+ snd_soc_dapm_nc_pin(codec, "VINL");
+ snd_soc_dapm_nc_pin(codec, "VINR");
+
+ /* Add magician specific controls */
+ err = snd_soc_add_controls(codec, uda1380_magician_controls,
+ ARRAY_SIZE(uda1380_magician_controls));
+ if (err < 0)
+ return err;
+
+ /* Add magician specific widgets */
+ snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+ ARRAY_SIZE(uda1380_dapm_widgets));
+
+ /* Set up magician specific audio path interconnects */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+/* magician digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link magician_dai[] = {
+{
+ .name = "uda1380",
+ .stream_name = "UDA1380 Playback",
+ .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1],
+ .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
+ .init = magician_uda1380_init,
+ .ops = &magician_playback_ops,
+},
+{
+ .name = "uda1380",
+ .stream_name = "UDA1380 Capture",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
+ .ops = &magician_capture_ops,
+}
+};
+
+/* magician audio machine driver */
+static struct snd_soc_card snd_soc_card_magician = {
+ .name = "Magician",
+ .dai_link = magician_dai,
+ .num_links = ARRAY_SIZE(magician_dai),
+ .platform = &pxa2xx_soc_platform,
+};
+
+/* magician audio private data */
+static struct uda1380_setup_data magician_uda1380_setup = {
+ .i2c_address = 0x18,
+ .dac_clk = UDA1380_DAC_CLK_WSPLL,
+};
+
+/* magician audio subsystem */
+static struct snd_soc_device magician_snd_devdata = {
+ .card = &snd_soc_card_magician,
+ .codec_dev = &soc_codec_dev_uda1380,
+ .codec_data = &magician_uda1380_setup,
+};
+
+static struct platform_device *magician_snd_device;
+
+static int __init magician_init(void)
+{
+ int ret;
+
+ if (!machine_is_magician())
+ return -ENODEV;
+
+ ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER");
+ if (ret)
+ goto err_request_power;
+ ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET");
+ if (ret)
+ goto err_request_reset;
+ ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
+ if (ret)
+ goto err_request_spk;
+ ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
+ if (ret)
+ goto err_request_ep;
+ ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
+ if (ret)
+ goto err_request_mic;
+ ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
+ if (ret)
+ goto err_request_in_sel0;
+ ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
+ if (ret)
+ goto err_request_in_sel1;
+
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1);
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
+
+ /* we may need to have the clock running here - pH5 */
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1);
+ udelay(5);
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0);
+
+ magician_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!magician_snd_device) {
+ ret = -ENOMEM;
+ goto err_pdev;
+ }
+
+ platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
+ magician_snd_devdata.dev = &magician_snd_device->dev;
+ ret = platform_device_add(magician_snd_device);
+ if (ret) {
+ platform_device_put(magician_snd_device);
+ goto err_pdev;
+ }
+
+ return 0;
+
+err_pdev:
+ gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+err_request_in_sel1:
+ gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+err_request_in_sel0:
+ gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+err_request_mic:
+ gpio_free(EGPIO_MAGICIAN_EP_POWER);
+err_request_ep:
+ gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+err_request_spk:
+ gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
+err_request_reset:
+ gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
+err_request_power:
+ return ret;
+}
+
+static void __exit magician_exit(void)
+{
+ platform_device_unregister(magician_snd_device);
+
+ gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0);
+
+ gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+ gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+ gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+ gpio_free(EGPIO_MAGICIAN_EP_POWER);
+ gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+ gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
+ gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
+}
+
+module_init(magician_init);
+module_exit(magician_exit);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC Magician");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 48a73f64500b..44fcc4e01e08 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -200,7 +200,7 @@ static struct snd_soc_device palm27x_snd_devdata = {
static struct platform_device *palm27x_snd_device;
-static int __init palm27x_asoc_init(void)
+static int palm27x_asoc_probe(struct platform_device *pdev)
{
int ret;
@@ -208,6 +208,10 @@ static int __init palm27x_asoc_init(void)
machine_is_palmld()))
return -ENODEV;
+ if (pdev->dev.platform_data)
+ palm27x_ep_gpio = ((struct palm27x_asoc_info *)
+ (pdev->dev.platform_data))->jack_gpio;
+
ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
if (ret)
return ret;
@@ -245,16 +249,31 @@ err_alloc:
return ret;
}
-static void __exit palm27x_asoc_exit(void)
+static int __devexit palm27x_asoc_remove(struct platform_device *pdev)
{
free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
gpio_free(palm27x_ep_gpio);
platform_device_unregister(palm27x_snd_device);
+ return 0;
}
-void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+static struct platform_driver palm27x_wm9712_driver = {
+ .probe = palm27x_asoc_probe,
+ .remove = __devexit_p(palm27x_asoc_remove),
+ .driver = {
+ .name = "palm27x-asoc",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init palm27x_asoc_init(void)
+{
+ return platform_driver_register(&palm27x_wm9712_driver);
+}
+
+static void __exit palm27x_asoc_exit(void)
{
- palm27x_ep_gpio = data->jack_gpio;
+ platform_driver_unregister(&palm27x_wm9712_driver);
}
module_init(palm27x_asoc_init);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 7acd3febf8b0..286be31545df 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -280,12 +280,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
* ssp_set_clkdiv - set SSP clock divider
* @div: serial clock rate divider
*/
-static void ssp_set_scr(struct ssp_dev *dev, u32 div)
+static void ssp_set_scr(struct ssp_device *ssp, u32 div)
{
- struct ssp_device *ssp = dev->ssp;
- u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+ u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+
+ if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+ sscr0 &= ~0x0000ff00;
+ sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
+ } else {
+ sscr0 &= ~0x000fff00;
+ sscr0 |= (div - 1) << 8; /* 1..4096 */
+ }
+ ssp_write_reg(ssp, SSCR0, sscr0);
+}
+
+/**
+ * ssp_get_clkdiv - get SSP clock divider
+ */
+static u32 ssp_get_scr(struct ssp_device *ssp)
+{
+ u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+ u32 div;
- ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div)));
+ if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+ div = ((sscr0 >> 8) & 0xff) * 2 + 2;
+ else
+ div = ((sscr0 >> 8) & 0xfff) + 1;
+ return div;
}
/*
@@ -326,7 +347,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
break;
case PXA_SSP_CLK_AUDIO:
priv->sysclk = 0;
- ssp_set_scr(&priv->dev, 1);
+ ssp_set_scr(ssp, 1);
sscr0 |= SSCR0_ACS;
break;
default:
@@ -387,7 +408,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
ssp_write_reg(ssp, SSACD, val);
break;
case PXA_SSP_DIV_SCR:
- ssp_set_scr(&priv->dev, div);
+ ssp_set_scr(ssp, div);
break;
default:
return -ENODEV;
@@ -627,12 +648,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
u32 sscr0;
u32 sspsp;
int width = snd_pcm_format_physical_width(params_format(params));
+ int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
/* select correct DMA params */
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
dma = 1; /* capture DMA offset is 1,3 */
- if (chn == 2)
- dma += 2; /* stereo DMA offset is 2, mono is 0 */
+ /* Network mode with one active slot (ttsa == 1) can be used
+ * to force 16-bit frame width on the wire (for S16_LE), even
+ * with two channels. Use 16-bit DMA transfers for this case.
+ */
+ if (((chn == 2) && (ttsa != 1)) || (width == 32))
+ dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */
+
cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
@@ -668,8 +695,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
case SND_SOC_DAIFMT_I2S:
sspsp = ssp_read_reg(ssp, SSPSP);
- if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) &&
- (width == 16)) {
+ if ((ssp_get_scr(ssp) == 4) && (width == 16)) {
/* This is a special case where the bitclk is 64fs
* and we're not dealing with 2*32 bits of audio
* samples.
@@ -712,7 +738,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
/* When we use a network mode, we always require TDM slots
* - complain loudly and fail if they've not been set up yet.
*/
- if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+ if ((sscr0 & SSCR0_MOD) && !ttsa) {
dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
return -EINVAL;
}
@@ -800,6 +826,7 @@ static int pxa_ssp_probe(struct platform_device *pdev,
goto err_priv;
}
+ priv->dai_fmt = (unsigned int) -1;
dai->private_data = priv;
return 0;
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 2f3a21eee051..df494d1e346f 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,10 +1,10 @@
config SND_S3C24XX_SOC
tristate "SoC Audio for the Samsung S3CXXXX chips"
- depends on ARCH_S3C2410 || ARCH_S3C64XX
+ depends on ARCH_S3C2410
help
Say Y or M if you want to add support for codecs attached to
- the S3C24XX and S3C64XX AC97, I2S or SSP interface. You will
- also need to select the audio interfaces to support below.
+ the S3C24XX AC97 or I2S interfaces. You will also need to
+ select the audio interfaces to support below.
config SND_S3C24XX_SOC_I2S
tristate
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 32063790d95b..93e6c87b7399 100644
--- a/sound/soc/s3c24xx/jive_wm8750.c
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -69,8 +69,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream,
break;
}
- s3c_i2sv2_calc_rate(&div, NULL, params_rate(params),
- s3c2412_get_iisclk());
+ s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
+ s3c2412_get_iisclk());
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
@@ -145,8 +145,9 @@ static struct snd_soc_dai_link jive_dai = {
};
/* jive audio machine driver */
-static struct snd_soc_machine snd_soc_machine_jive = {
+static struct snd_soc_card snd_soc_machine_jive = {
.name = "Jive",
+ .platform = &s3c24xx_soc_platform,
.dai_link = &jive_dai,
.num_links = 1,
};
@@ -157,9 +158,8 @@ static struct wm8750_setup_data jive_wm8750_setup = {
/* jive audio subsystem */
static struct snd_soc_device jive_snd_devdata = {
- .machine = &snd_soc_machine_jive,
- .platform = &s3c24xx_soc_platform,
- .codec_dev = &soc_codec_dev_wm8750_spi,
+ .card = &snd_soc_machine_jive,
+ .codec_dev = &soc_codec_dev_wm8750,
.codec_data = &jive_wm8750_setup,
};
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 295a4c910262..ab680aac3fcb 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -473,9 +473,9 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
/* default table of all avaialable root fs divisors */
static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 };
-int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
- unsigned int *fstab,
- unsigned int rate, struct clk *clk)
+int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+ unsigned int *fstab,
+ unsigned int rate, struct clk *clk)
{
unsigned long clkrate = clk_get_rate(clk);
unsigned int div;
@@ -531,7 +531,7 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
return 0;
}
-EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate);
int s3c_i2sv2_probe(struct platform_device *pdev,
struct snd_soc_dai *dai,
@@ -624,15 +624,18 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
{
- dai->ops.trigger = s3c2412_i2s_trigger;
- dai->ops.hw_params = s3c2412_i2s_hw_params;
- dai->ops.set_fmt = s3c2412_i2s_set_fmt;
- dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv;
+ struct snd_soc_dai_ops *ops = dai->ops;
+
+ ops->trigger = s3c2412_i2s_trigger;
+ ops->hw_params = s3c2412_i2s_hw_params;
+ ops->set_fmt = s3c2412_i2s_set_fmt;
+ ops->set_clkdiv = s3c2412_i2s_set_clkdiv;
dai->suspend = s3c2412_i2s_suspend;
dai->resume = s3c2412_i2s_resume;
return snd_soc_register_dai(dai);
}
-
EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 1ca3cdaa8213..b7e0b3f0bfc8 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -33,8 +33,8 @@
#include <plat/regs-s3c2412-iis.h>
-#include <plat/regs-gpio.h>
#include <plat/audio.h>
+#include <mach/regs-gpio.h>
#include <mach/dma.h>
#include "s3c24xx-pcm.h"
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6e710f705a74..99712f652d0d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -98,7 +98,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
int err;
codec->ac97->dev.bus = &ac97_bus_type;
- codec->ac97->dev.parent = NULL;
+ codec->ac97->dev.parent = codec->card->dev;
codec->ac97->dev.release = soc_ac97_device_release;
dev_set_name(&codec->ac97->dev, "%d-%d:%s",
@@ -767,11 +767,21 @@ static int soc_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
+ struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
- dev_dbg(socdev->dev, "scheduling resume work\n");
-
- if (!schedule_work(&card->deferred_resume_work))
- dev_err(socdev->dev, "resume work item may be lost\n");
+ /* AC97 devices might have other drivers hanging off them so
+ * need to resume immediately. Other drivers don't have that
+ * problem and may take a substantial amount of time to resume
+ * due to I/O costs and anti-pop so handle them out of line.
+ */
+ if (cpu_dai->ac97_control) {
+ dev_dbg(socdev->dev, "Resuming AC97 immediately\n");
+ soc_resume_deferred(&card->deferred_resume_work);
+ } else {
+ dev_dbg(socdev->dev, "Scheduling resume work\n");
+ if (!schedule_work(&card->deferred_resume_work))
+ dev_err(socdev->dev, "resume work item may be lost\n");
+ }
return 0;
}
Copyright © 1996 – 2023 Jason A. Donenfeld. All Rights Reverse Engineered.