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authorTakashi Iwai <tiwai@suse.de>2013-02-11 14:03:37 +0100
committerTakashi Iwai <tiwai@suse.de>2013-02-11 14:03:37 +0100
commit83048ea7b1d11f5e560dea53873fb51d860202eb (patch)
tree5cc41aa9c2999188d3148e8828d44c4724115f5e /sound
parente9a25e04b845aade311aaa268a696c5c4ff3eece (diff)
parent699ba4546cfe3f4eee73aff4f307bf16362f9232 (diff)
Merge tag 'asoc-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.9 A fairly quiet release for ASoC: - Support for a wider range of hardware in the compressed stream code. - The ability to mute capture streams as well as playback streams while inactive. - DT support for AK4642, FSI, Samsung I2S and WM8962. - AC'97 support for Tegra. - New driver for max98090, replacing the stub which was there. Due to dependencies we've also got support for asynchronous I/O in regmap and DTification of DMA support for Samsung platforms (used only by the I2S driver and SPI) merged here as well.
Diffstat (limited to 'sound')
-rw-r--r--sound/arm/pxa2xx-ac97-lib.c8
-rw-r--r--sound/soc/atmel/Kconfig6
-rw-r--r--sound/soc/atmel/atmel-pcm-pdc.c4
-rw-r--r--sound/soc/atmel/atmel-pcm.c2
-rw-r--r--sound/soc/atmel/atmel-pcm.h6
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c14
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c6
-rw-r--r--sound/soc/codecs/ak4642.c33
-rw-r--r--sound/soc/codecs/arizona.c247
-rw-r--r--sound/soc/codecs/arizona.h8
-rw-r--r--sound/soc/codecs/cs4271.c34
-rw-r--r--sound/soc/codecs/cs42l52.c4
-rwxr-xr-x[-rw-r--r--]sound/soc/codecs/max98090.c2685
-rwxr-xr-xsound/soc/codecs/max98090.h1549
-rw-r--r--sound/soc/codecs/tlv320aic3x.c87
-rw-r--r--sound/soc/codecs/tlv320aic3x.h4
-rw-r--r--sound/soc/codecs/tlv320dac33.c16
-rw-r--r--sound/soc/codecs/twl4030.c85
-rw-r--r--sound/soc/codecs/twl6040.c62
-rw-r--r--sound/soc/codecs/wm2000.c66
-rw-r--r--sound/soc/codecs/wm2000.h3
-rw-r--r--sound/soc/codecs/wm2200.c62
-rw-r--r--sound/soc/codecs/wm5100.c13
-rw-r--r--sound/soc/codecs/wm5102.c172
-rw-r--r--sound/soc/codecs/wm5110.c111
-rw-r--r--sound/soc/codecs/wm8350.c10
-rw-r--r--sound/soc/codecs/wm8804.c3
-rw-r--r--sound/soc/codecs/wm8962.c37
-rw-r--r--sound/soc/codecs/wm8983.c41
-rw-r--r--sound/soc/codecs/wm8985.c43
-rw-r--r--sound/soc/codecs/wm_adsp.c525
-rw-r--r--sound/soc/codecs/wm_adsp.h18
-rw-r--r--sound/soc/codecs/wmfw.h15
-rw-r--r--sound/soc/davinci/davinci-evm.c6
-rw-r--r--sound/soc/davinci/davinci-mcasp.c2
-rw-r--r--sound/soc/dwc/designware_i2s.c4
-rw-r--r--sound/soc/fsl/Kconfig9
-rw-r--r--sound/soc/fsl/Makefile5
-rw-r--r--sound/soc/fsl/imx-pcm-dma.c21
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c22
-rw-r--r--sound/soc/fsl/imx-pcm.c35
-rw-r--r--sound/soc/fsl/imx-pcm.h18
-rw-r--r--sound/soc/generic/simple-card.c63
-rw-r--r--sound/soc/mxs/mxs-saif.c49
-rw-r--r--sound/soc/omap/Kconfig19
-rw-r--r--sound/soc/omap/Makefile4
-rw-r--r--sound/soc/omap/n810.c4
-rw-r--r--sound/soc/omap/omap-hdmi.c2
-rw-r--r--sound/soc/omap/omap-pcm.c14
-rw-r--r--sound/soc/omap/omap-twl4030.c204
-rw-r--r--sound/soc/omap/omap3pandora.c8
-rw-r--r--sound/soc/omap/rx51.c8
-rw-r--r--sound/soc/omap/sdp3430.c278
-rw-r--r--sound/soc/omap/zoom2.c207
-rw-r--r--sound/soc/pxa/palm27x.c38
-rw-r--r--sound/soc/samsung/Kconfig6
-rw-r--r--sound/soc/samsung/dma.c3
-rw-r--r--sound/soc/samsung/dma.h1
-rw-r--r--sound/soc/samsung/i2s.c267
-rw-r--r--sound/soc/samsung/i2s.h7
-rw-r--r--sound/soc/samsung/s3c24xx-i2s.c2
-rw-r--r--sound/soc/samsung/smdk_wm8580.c7
-rw-r--r--sound/soc/samsung/smdk_wm8994.c30
-rw-r--r--sound/soc/sh/fsi.c242
-rw-r--r--sound/soc/soc-compress.c123
-rw-r--r--sound/soc/soc-core.c135
-rw-r--r--sound/soc/soc-dapm.c6
-rw-r--r--sound/soc/soc-pcm.c19
-rw-r--r--sound/soc/tegra/Kconfig19
-rw-r--r--sound/soc/tegra/Makefile4
-rw-r--r--sound/soc/tegra/tegra20_ac97.c480
-rw-r--r--sound/soc/tegra/tegra20_ac97.h95
-rw-r--r--sound/soc/tegra/tegra20_das.c13
-rw-r--r--sound/soc/tegra/tegra30_ahub.c4
-rw-r--r--sound/soc/tegra/tegra30_i2s.c4
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.c53
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.h1
-rw-r--r--sound/soc/tegra/tegra_wm9712.c176
-rw-r--r--sound/soc/ux500/mop500.c2
79 files changed, 6955 insertions, 1743 deletions
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index fff7753e35c1..e6f4633b8dd5 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -34,7 +34,7 @@ static struct clk *ac97_clk;
static struct clk *ac97conf_clk;
static int reset_gpio;
-extern void pxa27x_assert_ac97reset(int reset_gpio, int on);
+extern void pxa27x_configure_ac97reset(int reset_gpio, bool to_gpio);
/*
* Beware PXA27x bugs:
@@ -140,10 +140,10 @@ static inline void pxa_ac97_warm_pxa27x(void)
gsr_bits = 0;
/* warm reset broken on Bulverde, so manually keep AC97 reset high */
- pxa27x_assert_ac97reset(reset_gpio, 1);
+ pxa27x_configure_ac97reset(reset_gpio, true);
udelay(10);
GCR |= GCR_WARM_RST;
- pxa27x_assert_ac97reset(reset_gpio, 0);
+ pxa27x_configure_ac97reset(reset_gpio, false);
udelay(500);
}
@@ -358,7 +358,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev)
__func__, ret);
goto err_conf;
}
- pxa27x_assert_ac97reset(reset_gpio, 0);
+ pxa27x_configure_ac97reset(reset_gpio, false);
ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK");
if (IS_ERR(ac97conf_clk)) {
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index d1b691bf8e2d..3fdd87fa18a9 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -1,6 +1,6 @@
config SND_ATMEL_SOC
tristate "SoC Audio for the Atmel System-on-Chip"
- depends on ARCH_AT91
+ depends on HAS_IOMEM
help
Say Y or M if you want to add support for codecs attached to
the ATMEL SSC interface. You will also need
@@ -24,7 +24,7 @@ config SND_ATMEL_SOC_SSC
config SND_AT91_SOC_SAM9G20_WM8731
tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
- depends on ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS
+ depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS
select SND_ATMEL_SOC_PDC
select SND_ATMEL_SOC_SSC
select SND_SOC_WM8731
@@ -34,7 +34,7 @@ config SND_AT91_SOC_SAM9G20_WM8731
config SND_AT91_SOC_AFEB9260
tristate "SoC Audio support for AFEB9260 board"
- depends on ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
+ depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
select SND_ATMEL_SOC_PDC
select SND_ATMEL_SOC_SSC
select SND_SOC_TLV320AIC23
diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c
index 6a293c713a38..054ea4d9326a 100644
--- a/sound/soc/atmel/atmel-pcm-pdc.c
+++ b/sound/soc/atmel/atmel-pcm-pdc.c
@@ -159,7 +159,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
pr_debug("atmel-pcm: "
"hw_params: DMA for %s initialized "
- "(dma_bytes=%u, period_size=%u)\n",
+ "(dma_bytes=%zu, period_size=%zu)\n",
prtd->params->name,
runtime->dma_bytes,
prtd->period_size);
@@ -201,7 +201,7 @@ static int atmel_pcm_trigger(struct snd_pcm_substream *substream,
int ret = 0;
pr_debug("atmel-pcm:buffer_size = %ld,"
- "dma_area = %p, dma_bytes = %u\n",
+ "dma_area = %p, dma_bytes = %zu\n",
rtd->buffer_size, rtd->dma_area, rtd->dma_bytes);
switch (cmd) {
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index e99f1811300a..3109db7b9017 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -49,7 +49,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
buf->private_data = NULL;
buf->area = dma_alloc_coherent(pcm->card->dev, size,
&buf->addr, GFP_KERNEL);
- pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%d\n",
+ pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n",
(void *)buf->area, (void *)buf->addr, size);
if (!buf->area)
diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h
index bb45d20e7250..12ae814eff21 100644
--- a/sound/soc/atmel/atmel-pcm.h
+++ b/sound/soc/atmel/atmel-pcm.h
@@ -88,7 +88,8 @@ void atmel_pcm_free(struct snd_pcm *pcm);
int atmel_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma);
-#ifdef CONFIG_SND_ATMEL_SOC_PDC
+#if defined(CONFIG_SND_ATMEL_SOC_PDC) || \
+ defined(CONFIG_SND_ATMEL_SOC_PDC_MODULE)
int atmel_pcm_pdc_platform_register(struct device *dev);
void atmel_pcm_pdc_platform_unregister(struct device *dev);
#else
@@ -101,7 +102,8 @@ static inline void atmel_pcm_pdc_platform_unregister(struct device *dev)
}
#endif
-#ifdef CONFIG_SND_ATMEL_SOC_DMA
+#if defined(CONFIG_SND_ATMEL_SOC_DMA) || \
+ defined(CONFIG_SND_ATMEL_SOC_DMA_MODULE)
int atmel_pcm_dma_platform_register(struct device *dev);
void atmel_pcm_dma_platform_unregister(struct device *dev);
#else
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 1c7663422054..e13580d6c476 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -42,8 +42,6 @@
#include <sound/initval.h>
#include <sound/soc.h>
-#include <mach/hardware.h>
-
#include "atmel-pcm.h"
#include "atmel_ssc_dai.h"
@@ -679,15 +677,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
# define atmel_ssc_resume NULL
#endif /* CONFIG_PM */
-static int atmel_ssc_probe(struct snd_soc_dai *dai)
-{
- struct atmel_ssc_info *ssc_p = &ssc_info[dai->id];
-
- snd_soc_dai_set_drvdata(dai, ssc_p);
-
- return 0;
-}
-
#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000)
#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\
@@ -703,7 +692,6 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = {
};
static struct snd_soc_dai_driver atmel_ssc_dai = {
- .probe = atmel_ssc_probe,
.suspend = atmel_ssc_suspend,
.resume = atmel_ssc_resume,
.playback = {
@@ -790,8 +778,8 @@ void atmel_ssc_put_audio(int ssc_id)
{
struct ssc_device *ssc = ssc_info[ssc_id].ssc;
- ssc_free(ssc);
asoc_ssc_exit(&ssc->pdev->dev);
+ ssc_free(ssc);
}
EXPORT_SYMBOL_GPL(atmel_ssc_put_audio);
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index da976291da9e..2d6fbd0125b9 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -305,10 +305,10 @@ static int at91sam9g20ek_audio_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
- atmel_ssc_put_audio(0);
- snd_soc_unregister_card(card);
- clk_put(mclk);
+ clk_disable(mclk);
mclk = NULL;
+ snd_soc_unregister_card(card);
+ atmel_ssc_put_audio(0);
return 0;
}
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 1f0cdab03294..2d0378709702 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -26,6 +26,7 @@
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
+#include <linux/of_device.h>
#include <linux/module.h>
#include <sound/soc.h>
#include <sound/initval.h>
@@ -513,12 +514,31 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = {
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static struct of_device_id ak4642_of_match[];
static int ak4642_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
+ struct device_node *np = i2c->dev.of_node;
+ const struct snd_soc_codec_driver *driver;
+
+ driver = NULL;
+ if (np) {
+ const struct of_device_id *of_id;
+
+ of_id = of_match_device(ak4642_of_match, &i2c->dev);
+ if (of_id)
+ driver = of_id->data;
+ } else {
+ driver = (struct snd_soc_codec_driver *)id->driver_data;
+ }
+
+ if (!driver) {
+ dev_err(&i2c->dev, "no driver\n");
+ return -EINVAL;
+ }
+
return snd_soc_register_codec(&i2c->dev,
- (struct snd_soc_codec_driver *)id->driver_data,
- &ak4642_dai, 1);
+ driver, &ak4642_dai, 1);
}
static int ak4642_i2c_remove(struct i2c_client *client)
@@ -527,6 +547,14 @@ static int ak4642_i2c_remove(struct i2c_client *client)
return 0;
}
+static struct of_device_id ak4642_of_match[] = {
+ { .compatible = "asahi-kasei,ak4642", .data = &soc_codec_dev_ak4642},
+ { .compatible = "asahi-kasei,ak4643", .data = &soc_codec_dev_ak4642},
+ { .compatible = "asahi-kasei,ak4648", .data = &soc_codec_dev_ak4648},
+ {},
+};
+MODULE_DEVICE_TABLE(of, ak4642_of_match);
+
static const struct i2c_device_id ak4642_i2c_id[] = {
{ "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 },
{ "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 },
@@ -539,6 +567,7 @@ static struct i2c_driver ak4642_i2c_driver = {
.driver = {
.name = "ak4642-codec",
.owner = THIS_MODULE,
+ .of_match_table = ak4642_of_match,
},
.probe = ak4642_i2c_probe,
.remove = ak4642_i2c_remove,
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index ef62c435848e..d824c984c8a4 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -56,14 +56,14 @@
#define arizona_fll_warn(_fll, fmt, ...) \
dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
#define arizona_fll_dbg(_fll, fmt, ...) \
- dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+ dev_dbg(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
#define arizona_aif_err(_dai, fmt, ...) \
dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
#define arizona_aif_warn(_dai, fmt, ...) \
dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
#define arizona_aif_dbg(_dai, fmt, ...) \
- dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+ dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
"None",
@@ -141,6 +141,30 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
"ASRC1R",
"ASRC2L",
"ASRC2R",
+ "ISRC1INT1",
+ "ISRC1INT2",
+ "ISRC1INT3",
+ "ISRC1INT4",
+ "ISRC1DEC1",
+ "ISRC1DEC2",
+ "ISRC1DEC3",
+ "ISRC1DEC4",
+ "ISRC2INT1",
+ "ISRC2INT2",
+ "ISRC2INT3",
+ "ISRC2INT4",
+ "ISRC2DEC1",
+ "ISRC2DEC2",
+ "ISRC2DEC3",
+ "ISRC2DEC4",
+ "ISRC3INT1",
+ "ISRC3INT2",
+ "ISRC3INT3",
+ "ISRC3INT4",
+ "ISRC3DEC1",
+ "ISRC3DEC2",
+ "ISRC3DEC3",
+ "ISRC3DEC4",
};
EXPORT_SYMBOL_GPL(arizona_mixer_texts);
@@ -220,6 +244,30 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = {
0x91,
0x92,
0x93,
+ 0xa0, /* ISRC1INT1 */
+ 0xa1,
+ 0xa2,
+ 0xa3,
+ 0xa4, /* ISRC1DEC1 */
+ 0xa5,
+ 0xa6,
+ 0xa7,
+ 0xa8, /* ISRC2DEC1 */
+ 0xa9,
+ 0xaa,
+ 0xab,
+ 0xac, /* ISRC2INT1 */
+ 0xad,
+ 0xae,
+ 0xaf,
+ 0xb0, /* ISRC3DEC1 */
+ 0xb1,
+ 0xb2,
+ 0xb3,
+ 0xb4, /* ISRC3INT1 */
+ 0xb5,
+ 0xb6,
+ 0xb7,
};
EXPORT_SYMBOL_GPL(arizona_mixer_values);
@@ -275,6 +323,15 @@ const struct soc_enum arizona_lhpf4_mode =
arizona_lhpf_mode_text);
EXPORT_SYMBOL_GPL(arizona_lhpf4_mode);
+static const char *arizona_ng_hold_text[] = {
+ "30ms", "120ms", "250ms", "500ms",
+};
+
+const struct soc_enum arizona_ng_hold =
+ SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT,
+ 4, arizona_ng_hold_text);
+EXPORT_SYMBOL_GPL(arizona_ng_hold);
+
int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
int event)
{
@@ -417,6 +474,10 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
case 147456000:
val |= 6 << ARIZONA_SYSCLK_FREQ_SHIFT;
break;
+ case 0:
+ dev_dbg(arizona->dev, "%s cleared\n", name);
+ *clk = freq;
+ return 0;
default:
return -EINVAL;
}
@@ -635,6 +696,9 @@ static int arizona_startup(struct snd_pcm_substream *substream,
return 0;
}
+ if (base_rate == 0)
+ return 0;
+
if (base_rate % 8000)
constraint = &arizona_44k1_constraint;
else
@@ -645,25 +709,81 @@ static int arizona_startup(struct snd_pcm_substream *substream,
constraint);
}
+static int arizona_hw_params_rate(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+ int base = dai->driver->base;
+ int i, sr_val;
+
+ /*
+ * We will need to be more flexible than this in future,
+ * currently we use a single sample rate for SYSCLK.
+ */
+ for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++)
+ if (arizona_sr_vals[i] == params_rate(params))
+ break;
+ if (i == ARRAY_SIZE(arizona_sr_vals)) {
+ arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+ sr_val = i;
+
+ switch (dai_priv->clk) {
+ case ARIZONA_CLK_SYSCLK:
+ snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1,
+ ARIZONA_SAMPLE_RATE_1_MASK, sr_val);
+ if (base)
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
+ ARIZONA_AIF1_RATE_MASK, 0);
+ break;
+ case ARIZONA_CLK_ASYNCCLK:
+ snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
+ ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
+ if (base)
+ snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
+ ARIZONA_AIF1_RATE_MASK,
+ 8 << ARIZONA_AIF1_RATE_SHIFT);
+ break;
+ default:</