Merge branch 'for-3.0' into for-3.1
diff --git a/MAINTAINERS b/MAINTAINERS
index 29801f7..7bcc395 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -533,6 +533,8 @@
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
W: http://wiki.analog.com/
S: Supported
+F: sound/soc/codecs/adau*
+F: sound/soc/codecs/adav*
F: sound/soc/codecs/ad1*
F: sound/soc/codecs/ssm*
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 1bafe95..5ad5f3a 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -209,6 +209,10 @@
struct snd_soc_pcm_stream capture;
struct snd_soc_pcm_stream playback;
unsigned int symmetric_rates:1;
+
+ /* probe ordering - for components with runtime dependencies */
+ int probe_order;
+ int remove_order;
};
/*
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c46e7d8..602024d 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -348,6 +348,8 @@
void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
+int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route, int num);
/* dapm events */
int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
@@ -429,6 +431,7 @@
/* status */
u32 connect:1; /* source and sink widgets are connected */
u32 walked:1; /* path has been walked */
+ u32 weak:1; /* path ignored for power management */
int (*connected)(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink);
@@ -510,7 +513,7 @@
struct snd_soc_card *card; /* parent card */
/* used during DAPM updates */
- int dev_power;
+ enum snd_soc_bias_level target_bias_level;
struct list_head list;
#ifdef CONFIG_DEBUG_FS
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 3a4bd3a..6424b10 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -203,6 +203,16 @@
SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
/*
+ * Component probe and remove ordering levels for components with runtime
+ * dependencies.
+ */
+#define SND_SOC_COMP_ORDER_FIRST -2
+#define SND_SOC_COMP_ORDER_EARLY -1
+#define SND_SOC_COMP_ORDER_NORMAL 0
+#define SND_SOC_COMP_ORDER_LATE 1
+#define SND_SOC_COMP_ORDER_LAST 2
+
+/*
* Bias levels
*
* @ON: Bias is fully on for audio playback and capture operations.
@@ -214,10 +224,10 @@
* @OFF: Power Off. No restrictions on transition times.
*/
enum snd_soc_bias_level {
- SND_SOC_BIAS_OFF,
- SND_SOC_BIAS_STANDBY,
- SND_SOC_BIAS_PREPARE,
- SND_SOC_BIAS_ON,
+ SND_SOC_BIAS_OFF = 0,
+ SND_SOC_BIAS_STANDBY = 1,
+ SND_SOC_BIAS_PREPARE = 2,
+ SND_SOC_BIAS_ON = 3,
};
struct snd_jack;
@@ -258,6 +268,11 @@
SND_SOC_RBTREE_COMPRESSION
};
+enum snd_soc_pcm_subclass {
+ SND_SOC_PCM_CLASS_PCM = 0,
+ SND_SOC_PCM_CLASS_BE = 1,
+};
+
int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
unsigned int freq, int dir);
int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
@@ -612,6 +627,10 @@
void (*seq_notifier)(struct snd_soc_dapm_context *,
enum snd_soc_dapm_type, int);
+
+ /* probe ordering - for components with runtime dependencies */
+ int probe_order;
+ int remove_order;
};
/* SoC platform interface */
@@ -623,8 +642,7 @@
int (*resume)(struct snd_soc_dai *dai);
/* pcm creation and destruction */
- int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
- struct snd_pcm *);
+ int (*pcm_new)(struct snd_soc_pcm_runtime *);
void (*pcm_free)(struct snd_pcm *);
/*
@@ -636,6 +654,10 @@
/* platform stream ops */
struct snd_pcm_ops *ops;
+
+ /* probe ordering - for components with runtime dependencies */
+ int probe_order;
+ int remove_order;
};
struct snd_soc_platform {
@@ -725,8 +747,10 @@
/* callbacks */
int (*set_bias_level)(struct snd_soc_card *,
+ struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level);
int (*set_bias_level_post)(struct snd_soc_card *,
+ struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level);
long pmdown_time;
@@ -789,6 +813,9 @@
struct device dev;
struct snd_soc_card *card;
struct snd_soc_dai_link *dai_link;
+ struct mutex pcm_mutex;
+ enum snd_soc_pcm_subclass pcm_subclass;
+ struct snd_pcm_ops ops;
unsigned int complete:1;
unsigned int dev_registered:1;
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 1ed61c5..4f91387 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,5 @@
snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
+snd-soc-core-objs += soc-pcm.o soc-io.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index d0e7532..f81d4c3 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -364,9 +364,11 @@
\*--------------------------------------------------------------------------*/
static u64 atmel_pcm_dmamask = 0xffffffff;
-static int atmel_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
@@ -382,7 +384,7 @@
}
if (dai->driver->capture.channels_min) {
- pr_debug("at32-pcm:"
+ pr_debug("atmel-pcm:"
"Allocating PCM capture DMA buffer\n");
ret = atmel_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE);
diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h
index 2597329..5e0a95e 100644
--- a/sound/soc/atmel/atmel-pcm.h
+++ b/sound/soc/atmel/atmel-pcm.h
@@ -60,7 +60,7 @@
* This structure, shared between the PCM driver and the interface,
* contains all information required by the PCM driver to perform the
* PDC DMA operation. All fields except dma_intr_handler() are initialized
- * by the interface. The dms_intr_handler() pointer is set by the PCM
+ * by the interface. The dma_intr_handler() pointer is set by the PCM
* driver and called by the interface SSC interrupt handler if it is
* non-NULL.
*/
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index eda955b..7122509 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -402,7 +402,7 @@
if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
&& bits > 16) {
printk(KERN_WARNING
- "atmel_ssc_dai: sample size %d"
+ "atmel_ssc_dai: sample size %d "
"is too large for I2S\n", bits);
return -EINVAL;
}
@@ -838,10 +838,8 @@
}
ssc_pdev = platform_device_alloc("atmel-ssc-dai", ssc_id);
- if (!ssc_pdev) {
- ssc_free(ssc);
+ if (!ssc_pdev)
return -ENOMEM;
- }
/* If we can grab the SSC briefly to parent the DAI device off it */
ssc = ssc_request(ssc_id);
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 95572d2..bad3aa1 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -92,6 +92,7 @@
};
static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
static int mclk_on;
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 10fdd28..20bb53a 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -319,10 +319,11 @@
snd_pcm_lib_preallocate_free_for_all(pcm);
}
-static int au1xpsc_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index ae40359..fe9d548 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -10,13 +10,36 @@
config SND_BF5XX_SOC_SSM2602
tristate "SoC SSM2602 Audio support for BF52x ezkit"
- depends on SND_BF5XX_I2S
+ depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
select SND_BF5XX_SOC_I2S
select SND_SOC_SSM2602
- select I2C
help
Say Y if you want to add support for SoC audio on BF527-EZKIT.
+config SND_SOC_BFIN_EVAL_ADAU1701
+ tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards"
+ depends on SND_BF5XX_I2S
+ select SND_BF5XX_SOC_I2S
+ select SND_SOC_ADAU1701
+ select I2C
+ help
+ Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ
+ board connected to one of the Blackfin evaluation boards like the
+ BF5XX-STAMP or BF5XX-EZKIT.
+
+config SND_SOC_BFIN_EVAL_ADAV80X
+ tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
+ depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+ select SND_BF5XX_SOC_I2S
+ select SND_SOC_ADAV80X
+ help
+ Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or
+ EVAL-ADAV803 board connected to one of the Blackfin evaluation boards
+ like the BF5XX-STAMP or BF5XX-EZKIT.
+
+ Note: This driver assumes that the ADAV80X digital record and playback
+ interfaces are connected to the first SPORT port on the BF5XX board.
+
config SND_BF5XX_SOC_AD73311
tristate "SoC AD73311 Audio support for Blackfin"
depends on SND_BF5XX_I2S
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 49af3f3..6018bf5 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -21,9 +21,13 @@
snd-ssm2602-objs := bf5xx-ssm2602.o
snd-ad73311-objs := bf5xx-ad73311.o
snd-ad193x-objs := bf5xx-ad193x.o
+snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o
+snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o
obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o
obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 98b44b3..9e59f68 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -418,9 +418,11 @@
static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
-int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
pr_debug("%s enter\n", __func__);
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index f1fd95b..61ddf94 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -168,7 +168,7 @@
snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware);
- ret = snd_pcm_hw_constraint_integer(runtime, \
+ ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
goto out;
@@ -257,9 +257,11 @@
static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
-int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
pr_debug("%s enter\n", __func__);
@@ -304,8 +306,8 @@
static struct platform_driver bfin_i2s_pcm_driver = {
.driver = {
- .name = "bfin-i2s-pcm-audio",
- .owner = THIS_MODULE,
+ .name = "bfin-i2s-pcm-audio",
+ .owner = THIS_MODULE,
},
.probe = bfin_i2s_soc_platform_probe,
diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c
index 07cfc7a..c95cc03 100644
--- a/sound/soc/blackfin/bf5xx-tdm-pcm.c
+++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c
@@ -283,9 +283,11 @@
static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
-static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c
new file mode 100644
index 0000000..e5550ac
--- /dev/null
+++ b/sound/soc/blackfin/bfin-eval-adau1701.c
@@ -0,0 +1,139 @@
+/*
+ * Machine driver for EVAL-ADAU1701MINIZ on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/adau1701.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adau1701_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adau1701_dapm_routes[] = {
+ { "Speaker", NULL, "OUT0" },
+ { "Speaker", NULL, "OUT1" },
+ { "Line Out", NULL, "OUT2" },
+ { "Line Out", NULL, "OUT3" },
+
+ { "IN0", NULL, "Line In" },
+ { "IN1", NULL, "Line In" },
+};
+
+static int bfin_eval_adau1701_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000,
+ SND_SOC_CLOCK_IN);
+
+ return ret;
+}
+
+static struct snd_soc_ops bfin_eval_adau1701_ops = {
+ .hw_params = bfin_eval_adau1701_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = {
+ {
+ .name = "adau1701",
+ .stream_name = "adau1701",
+ .cpu_dai_name = "bfin-i2s.0",
+ .codec_dai_name = "adau1701",
+ .platform_name = "bfin-i2s-pcm-audio",
+ .codec_name = "adau1701.0-0034",
+ .ops = &bfin_eval_adau1701_ops,
+ },
+ {
+ .name = "adau1701",
+ .stream_name = "adau1701",
+ .cpu_dai_name = "bfin-i2s.1",
+ .codec_dai_name = "adau1701",
+ .platform_name = "bfin-i2s-pcm-audio",
+ .codec_name = "adau1701.0-0034",
+ .ops = &bfin_eval_adau1701_ops,
+ },
+};
+
+static struct snd_soc_card bfin_eval_adau1701 = {
+ .name = "bfin-eval-adau1701",
+ .dai_link = &bfin_eval_adau1701_dai[CONFIG_SND_BF5XX_SPORT_NUM],
+ .num_links = 1,
+
+ .dapm_widgets = bfin_eval_adau1701_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1701_dapm_widgets),
+ .dapm_routes = bfin_eval_adau1701_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1701_dapm_routes),
+};
+
+static int bfin_eval_adau1701_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &bfin_eval_adau1701;
+
+ card->dev = &pdev->dev;
+
+ return snd_soc_register_card(&bfin_eval_adau1701);
+}
+
+static int __devexit bfin_eval_adau1701_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver bfin_eval_adau1701_driver = {
+ .driver = {
+ .name = "bfin-eval-adau1701",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = bfin_eval_adau1701_probe,
+ .remove = __devexit_p(bfin_eval_adau1701_remove),
+};
+
+static int __init bfin_eval_adau1701_init(void)
+{
+ return platform_driver_register(&bfin_eval_adau1701_driver);
+}
+module_init(bfin_eval_adau1701_init);
+
+static void __exit bfin_eval_adau1701_exit(void)
+{
+ platform_driver_unregister(&bfin_eval_adau1701_driver);
+}
+module_exit(bfin_eval_adau1701_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:bfin-eval-adau1701");
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
new file mode 100644
index 0000000..8d014d0
--- /dev/null
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -0,0 +1,173 @@
+/*
+ * Machine driver for EVAL-ADAV801 and EVAL-ADAV803 on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include "../codecs/adav80x.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adav80x_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adav80x_dapm_routes[] = {
+ { "Line Out", NULL, "VOUTL" },
+ { "Line Out", NULL, "VOUTR" },
+
+ { "VINL", NULL, "Line In" },
+ { "VINR", NULL, "Line In" },
+};
+
+static int bfin_eval_adav80x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL,
+ 27000000, params_rate(params) * 256);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_PLL1,
+ params_rate(params) * 256, SND_SOC_CLOCK_IN);
+
+ return ret;
+}
+
+static int bfin_eval_adav80x_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK1, 0,
+ SND_SOC_CLOCK_OUT);
+ snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK2, 0,
+ SND_SOC_CLOCK_OUT);
+ snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK3, 0,
+ SND_SOC_CLOCK_OUT);
+
+ snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_XTAL, 2700000, 0);
+
+ return 0;
+}
+
+static struct snd_soc_ops bfin_eval_adav80x_ops = {
+ .hw_params = bfin_eval_adav80x_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = {
+ {
+ .name = "adav80x",
+ .stream_name = "ADAV80x HiFi",
+ .cpu_dai_name = "bfin-i2s.0",
+ .codec_dai_name = "adav80x-hifi",
+ .platform_name = "bfin-i2s-pcm-audio",
+ .init = bfin_eval_adav80x_codec_init,
+ .ops = &bfin_eval_adav80x_ops,
+ },
+};
+
+static struct snd_soc_card bfin_eval_adav80x = {
+ .name = "bfin-eval-adav80x",
+ .dai_link = bfin_eval_adav80x_dais,
+ .num_links = ARRAY_SIZE(bfin_eval_adav80x_dais),
+
+ .dapm_widgets = bfin_eval_adav80x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adav80x_dapm_widgets),
+ .dapm_routes = bfin_eval_adav80x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(bfin_eval_adav80x_dapm_routes),
+};
+
+enum bfin_eval_adav80x_type {
+ BFIN_EVAL_ADAV801,
+ BFIN_EVAL_ADAV803,
+};
+
+static int bfin_eval_adav80x_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &bfin_eval_adav80x;
+ const char *codec_name;
+
+ switch (platform_get_device_id(pdev)->driver_data) {
+ case BFIN_EVAL_ADAV801:
+ codec_name = "spi0.1";
+ break;
+ case BFIN_EVAL_ADAV803:
+ codec_name = "adav803.0-0034";
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ bfin_eval_adav80x_dais[0].codec_name = codec_name;
+
+ card->dev = &pdev->dev;
+
+ return snd_soc_register_card(&bfin_eval_adav80x);
+}
+
+static int __devexit bfin_eval_adav80x_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static const struct platform_device_id bfin_eval_adav80x_ids[] = {
+ { "bfin-eval-adav801", BFIN_EVAL_ADAV801 },
+ { "bfin-eval-adav803", BFIN_EVAL_ADAV803 },
+ { },
+};
+MODULE_DEVICE_TABLE(platform, bfin_eval_adav80x_ids);
+
+static struct platform_driver bfin_eval_adav80x_driver = {
+ .driver = {
+ .name = "bfin-eval-adav80x",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = bfin_eval_adav80x_probe,
+ .remove = __devexit_p(bfin_eval_adav80x_remove),
+ .id_table = bfin_eval_adav80x_ids,
+};
+
+static int __init bfin_eval_adav80x_init(void)
+{
+ return platform_driver_register(&bfin_eval_adav80x_driver);
+}
+module_init(bfin_eval_adav80x_init);
+
+static void __exit bfin_eval_adav80x_exit(void)
+{
+ platform_driver_unregister(&bfin_eval_adav80x_driver);
+}
+module_exit(bfin_eval_adav80x_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 98175a0..ff43405 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -17,6 +17,7 @@
select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
select SND_SOC_AD1980 if SND_SOC_AC97_BUS
select SND_SOC_AD73311
+ select SND_SOC_ADAV80X
select SND_SOC_ADS117X
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
@@ -42,6 +43,7 @@
select SND_SOC_SN95031 if INTEL_SCU_IPC
select SND_SOC_SPDIF
select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_STA32X if I2C
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TLV320AIC23 if I2C
select SND_SOC_TLV320AIC26 if SPI_MASTER
@@ -71,6 +73,7 @@
select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8770 if SPI_MASTER
select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8782
select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8900 if I2C
select SND_SOC_WM8903 if I2C
@@ -130,7 +133,14 @@
config SND_SOC_AD73311
tristate
-
+
+config SND_SOC_ADAU1701
+ select SIGMA
+ tristate
+
+config SND_SOC_ADAV80X
+ tristate
+
config SND_SOC_ADS117X
tristate
@@ -216,6 +226,9 @@
config SND_SOC_SSM2602
tristate
+config SND_SOC_STA32X
+ tristate
+
config SND_SOC_STAC9766
tristate
@@ -299,6 +312,9 @@
config SND_SOC_WM8776
tristate
+config SND_SOC_WM8782
+ tristate
+
config SND_SOC_WM8804
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index fd85584..4957431 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -4,6 +4,8 @@
snd-soc-ad193x-objs := ad193x.o
snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
+snd-soc-adau1701-objs := adau1701.o
+snd-soc-adav80x-objs := adav80x.o
snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
@@ -28,6 +30,7 @@
snd-soc-sn95031-objs := sn95031.o
snd-soc-spdif-objs := spdif_transciever.o
snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-sta32x-objs := sta32x.o
snd-soc-stac9766-objs := stac9766.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
@@ -55,6 +58,7 @@
snd-soc-wm8753-objs := wm8753.o
snd-soc-wm8770-objs := wm8770.o
snd-soc-wm8776-objs := wm8776.o
+snd-soc-wm8782-objs := wm8782.o
snd-soc-wm8804-objs := wm8804.o
snd-soc-wm8900-objs := wm8900.o
snd-soc-wm8903-objs := wm8903.o
@@ -95,6 +99,8 @@
obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o
+obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o
obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
@@ -120,6 +126,7 @@
obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o
obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
@@ -147,6 +154,7 @@
obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o
obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o
+obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o
obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o
obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 754c496..4e5c572 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -1,19 +1,10 @@
-/*
- * File: sound/soc/codecs/ad1836.c
- * Author: Barry Song <Barry.Song@analog.com>
+ /*
+ * Audio Codec driver supporting:
+ * AD1835A, AD1836, AD1837A, AD1838A, AD1839A
*
- * Created: Aug 04 2009
- * Description: Driver for AD1836 sound chip
+ * Copyright 2009-2011 Analog Devices Inc.
*
- * Modified:
- * Copyright 2009 Analog Devices Inc.
- *
- * Bugs: Enter bugs at http://blackfin.uclinux.org/
- *
- * 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.
+ * Licensed under the GPL-2 or later.
*/
#include <linux/init.h>
@@ -30,10 +21,15 @@
#include <linux/spi/spi.h>
#include "ad1836.h"
+enum ad1836_type {
+ AD1835,
+ AD1836,
+ AD1838,
+};
+
/* codec private data */
struct ad1836_priv {
- enum snd_soc_control_type control_type;
- void *control_data;
+ enum ad1836_type type;
};
/*
@@ -44,29 +40,60 @@
static const struct soc_enum ad1836_deemp_enum =
SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp);
-static const struct snd_kcontrol_new ad1836_snd_controls[] = {
- /* DAC volume control */
- SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL,
- AD1836_DAC_R1_VOL, 0, 0x3FF, 0),
- SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL,
- AD1836_DAC_R2_VOL, 0, 0x3FF, 0),
- SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL,
- AD1836_DAC_R3_VOL, 0, 0x3FF, 0),
+#define AD1836_DAC_VOLUME(x) \
+ SOC_DOUBLE_R("DAC" #x " Playback Volume", AD1836_DAC_L_VOL(x), \
+ AD1836_DAC_R_VOL(x), 0, 0x3FF, 0)
- /* ADC switch control */
- SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE,
- AD1836_ADCR1_MUTE, 1, 1),
- SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE,
- AD1836_ADCR2_MUTE, 1, 1),
+#define AD1836_DAC_SWITCH(x) \
+ SOC_DOUBLE("DAC" #x " Playback Switch", AD1836_DAC_CTRL2, \
+ AD1836_MUTE_LEFT(x), AD1836_MUTE_RIGHT(x), 1, 1)
- /* DAC switch control */
- SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE,
- AD1836_DACR1_MUTE, 1, 1),
- SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE,
- AD1836_DACR2_MUTE, 1, 1),
- SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE,
- AD1836_DACR3_MUTE, 1, 1),
+#define AD1836_ADC_SWITCH(x) \
+ SOC_DOUBLE("ADC" #x " Capture Switch", AD1836_ADC_CTRL2, \
+ AD1836_MUTE_LEFT(x), AD1836_MUTE_RIGHT(x), 1, 1)
+static const struct snd_kcontrol_new ad183x_dac_controls[] = {
+ AD1836_DAC_VOLUME(1),
+ AD1836_DAC_SWITCH(1),
+ AD1836_DAC_VOLUME(2),
+ AD1836_DAC_SWITCH(2),
+ AD1836_DAC_VOLUME(3),
+ AD1836_DAC_SWITCH(3),
+ AD1836_DAC_VOLUME(4),
+ AD1836_DAC_SWITCH(4),
+};
+
+static const struct snd_soc_dapm_widget ad183x_dac_dapm_widgets[] = {
+ SND_SOC_DAPM_OUTPUT("DAC1OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC2OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC3OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC4OUT"),
+};
+
+static const struct snd_soc_dapm_route ad183x_dac_routes[] = {
+ { "DAC1OUT", NULL, "DAC" },
+ { "DAC2OUT", NULL, "DAC" },
+ { "DAC3OUT", NULL, "DAC" },
+ { "DAC4OUT", NULL, "DAC" },
+};
+
+static const struct snd_kcontrol_new ad183x_adc_controls[] = {
+ AD1836_ADC_SWITCH(1),
+ AD1836_ADC_SWITCH(2),
+ AD1836_ADC_SWITCH(3),
+};
+
+static const struct snd_soc_dapm_widget ad183x_adc_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("ADC1IN"),
+ SND_SOC_DAPM_INPUT("ADC2IN"),
+};
+
+static const struct snd_soc_dapm_route ad183x_adc_routes[] = {
+ { "ADC", NULL, "ADC1IN" },
+ { "ADC", NULL, "ADC2IN" },
+};
+
+static const struct snd_kcontrol_new ad183x_controls[] = {
/* ADC high-pass filter */
SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1,
AD1836_ADC_HIGHPASS_FILTER, 1, 0),
@@ -75,27 +102,24 @@
SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum),
};
-static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget ad183x_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1,
AD1836_DAC_POWERDOWN, 1),
SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1,
AD1836_ADC_POWERDOWN, 1, NULL, 0),
- SND_SOC_DAPM_OUTPUT("DAC1OUT"),
- SND_SOC_DAPM_OUTPUT("DAC2OUT"),
- SND_SOC_DAPM_OUTPUT("DAC3OUT"),
- SND_SOC_DAPM_INPUT("ADC1IN"),
- SND_SOC_DAPM_INPUT("ADC2IN"),
};
-static const struct snd_soc_dapm_route audio_paths[] = {
+static const struct snd_soc_dapm_route ad183x_dapm_routes[] = {
{ "DAC", NULL, "ADC_PWR" },
{ "ADC", NULL, "ADC_PWR" },
- { "DAC1OUT", "DAC1 Switch", "DAC" },
- { "DAC2OUT", "DAC2 Switch", "DAC" },
- { "DAC3OUT", "DAC3 Switch", "DAC" },
- { "ADC", "ADC1 Switch", "ADC1IN" },
- { "ADC", "ADC2 Switch", "ADC2IN" },
+};
+
+static const DECLARE_TLV_DB_SCALE(ad1836_in_tlv, 0, 300, 0);
+
+static const struct snd_kcontrol_new ad1836_controls[] = {
+ SOC_DOUBLE_TLV("ADC2 Capture Volume", AD1836_ADC_CTRL1, 3, 0, 4, 0,
+ ad1836_in_tlv),
};
/*
@@ -165,64 +189,69 @@
return 0;
}
-#ifdef CONFIG_PM
-static int ad1836_soc_suspend(struct snd_soc_codec *codec,
- pm_message_t state)
-{
- /* reset clock control mode */
- u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
- adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK;
-
- return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
-}
-
-static int ad1836_soc_resume(struct snd_soc_codec *codec)
-{
- /* restore clock control mode */
- u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
- adc_ctrl2 |= AD1836_ADC_AUX;
-
- return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
-}
-#else
-#define ad1836_soc_suspend NULL
-#define ad1836_soc_resume NULL
-#endif
-
static struct snd_soc_dai_ops ad1836_dai_ops = {
.hw_params = ad1836_hw_params,
.set_fmt = ad1836_set_dai_fmt,
};
-/* codec DAI instance */
-static struct snd_soc_dai_driver ad1836_dai = {
- .name = "ad1836-hifi",
- .playback = {
- .stream_name = "Playback",
- .channels_min = 2,
- .channels_max = 6,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
- },
- .capture = {
- .stream_name = "Capture",
- .channels_min = 2,
- .channels_max = 4,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
- },
- .ops = &ad1836_dai_ops,
+#define AD183X_DAI(_name, num_dacs, num_adcs) \
+{ \
+ .name = _name "-hifi", \
+ .playback = { \
+ .stream_name = "Playback", \
+ .channels_min = 2, \
+ .channels_max = (num_dacs) * 2, \
+ .rates = SNDRV_PCM_RATE_48000, \
+ .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, \
+ }, \
+ .capture = { \
+ .stream_name = "Capture", \
+ .channels_min = 2, \
+ .channels_max = (num_adcs) * 2, \
+ .rates = SNDRV_PCM_RATE_48000, \
+ .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, \
+ }, \
+ .ops = &ad1836_dai_ops, \
+}
+
+static struct snd_soc_dai_driver ad183x_dais[] = {
+ [AD1835] = AD183X_DAI("ad1835", 4, 1),
+ [AD1836] = AD183X_DAI("ad1836", 3, 2),
+ [AD1838] = AD183X_DAI("ad1838", 3, 1),
};
+#ifdef CONFIG_PM
+static int ad1836_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ /* reset clock control mode */
+ return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
+ AD1836_ADC_SERFMT_MASK, 0);
+}
+
+static int ad1836_resume(struct snd_soc_codec *codec)
+{
+ /* restore clock control mode */
+ return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
+ AD1836_ADC_SERFMT_MASK, AD1836_ADC_AUX);
+}
+#else
+#define ad1836_suspend NULL
+#define ad1836_resume NULL
+#endif
+
static int ad1836_probe(struct snd_soc_codec *codec)
{
struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int num_dacs, num_adcs;
int ret = 0;
+ int i;
- codec->control_data = ad1836->control_data;
+ num_dacs = ad183x_dais[ad1836->type].playback.channels_max / 2;
+ num_adcs = ad183x_dais[ad1836->type].capture.channels_max / 2;
+
ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI);
if (ret < 0) {
dev_err(codec->dev, "failed to set cache I/O: %d\n",
@@ -239,21 +268,46 @@
snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100);
/* unmute adc channles, adc aux mode */
snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180);
- /* left/right diff:PGA/MUX */
- snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
/* volume */
- snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF);
- snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF);
- snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF);
- snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF);
- snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF);
- snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF);
+ for (i = 1; i <= num_dacs; ++i) {
+ snd_soc_write(codec, AD1836_DAC_L_VOL(i), 0x3FF);
+ snd_soc_write(codec, AD1836_DAC_R_VOL(i), 0x3FF);
+ }
- snd_soc_add_controls(codec, ad1836_snd_controls,
- ARRAY_SIZE(ad1836_snd_controls));
- snd_soc_dapm_new_controls(dapm, ad1836_dapm_widgets,
- ARRAY_SIZE(ad1836_dapm_widgets));
- snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
+ if (ad1836->type == AD1836) {
+ /* left/right diff:PGA/MUX */
+ snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
+ ret = snd_soc_add_controls(codec, ad1836_controls,
+ ARRAY_SIZE(ad1836_controls));
+ if (ret)
+ return ret;
+ } else {
+ snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00);
+ }
+
+ ret = snd_soc_add_controls(codec, ad183x_dac_controls, num_dacs * 2);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_add_controls(codec, ad183x_adc_controls, num_adcs);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_new_controls(dapm, ad183x_dac_dapm_widgets, num_dacs);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_new_controls(dapm, ad183x_adc_dapm_widgets, num_adcs);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_add_routes(dapm, ad183x_dac_routes, num_dacs);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_add_routes(dapm, ad183x_adc_routes, num_adcs);
+ if (ret)
+ return ret;
return ret;
}
@@ -262,19 +316,24 @@
static int ad1836_remove(struct snd_soc_codec *codec)
{
/* reset clock control mode */
- u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
- adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK;
-
- return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
+ return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
+ AD1836_ADC_SERFMT_MASK, 0);
}
static struct snd_soc_codec_driver soc_codec_dev_ad1836 = {
- .probe = ad1836_probe,
- .remove = ad1836_remove,
- .suspend = ad1836_soc_suspend,
- .resume = ad1836_soc_resume,
+ .probe = ad1836_probe,
+ .remove = ad1836_remove,
+ .suspend = ad1836_suspend,
+ .resume = ad1836_resume,
.reg_cache_size = AD1836_NUM_REGS,
.reg_word_size = sizeof(u16),
+
+ .controls = ad183x_controls,
+ .num_controls = ARRAY_SIZE(ad183x_controls),
+ .dapm_widgets = ad183x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ad183x_dapm_widgets),
+ .dapm_routes = ad183x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(ad183x_dapm_routes),
};
static int __devinit ad1836_spi_probe(struct spi_device *spi)
@@ -286,12 +345,12 @@
if (ad1836 == NULL)
return -ENOMEM;
+ ad1836->type = spi_get_device_id(spi)->driver_data;
+
spi_set_drvdata(spi, ad1836);
- ad1836->control_data = spi;
- ad1836->control_type = SND_SOC_SPI;
ret = snd_soc_register_codec(&spi->dev,
- &soc_codec_dev_ad1836, &ad1836_dai, 1);
+ &soc_codec_dev_ad1836, &ad183x_dais[ad1836->type], 1);
if (ret < 0)
kfree(ad1836);
return ret;
@@ -303,27 +362,29 @@
kfree(spi_get_drvdata(spi));
return 0;
}
+static const struct spi_device_id ad1836_ids[] = {
+ { "ad1835", AD1835 },
+ { "ad1836", AD1836 },
+ { "ad1837", AD1835 },
+ { "ad1838", AD1838 },
+ { "ad1839", AD1838 },
+ { },
+};
+MODULE_DEVICE_TABLE(spi, ad1836_ids);
static struct spi_driver ad1836_spi_driver = {
.driver = {
- .name = "ad1836-codec",
+ .name = "ad1836",
.owner = THIS_MODULE,
},
.probe = ad1836_spi_probe,
.remove = __devexit_p(ad1836_spi_remove),
+ .id_table = ad1836_ids,
};
static int __init ad1836_init(void)
{
- int ret;
-
- ret = spi_register_driver(&ad1836_spi_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n",
- ret);
- }
-
- return ret;
+ return spi_register_driver(&ad1836_spi_driver);
}
module_init(ad1836_init);
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 9d6a3f8..444747f 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -1,19 +1,10 @@
/*
- * File: sound/soc/codecs/ad1836.h
- * Based on:
- * Author: Barry Song <Barry.Song@analog.com>
+ * Audio Codec driver supporting:
+ * AD1835A, AD1836, AD1837A, AD1838A, AD1839A
*
- * Created: Aug 04, 2009
- * Description: definitions for AD1836 registers
+ * Copyright 2009-2011 Analog Devices Inc.
*
- * Modified:
- *
- * Bugs: Enter bugs at http://blackfin.uclinux.org/
- *
- * 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.
+ * Licensed under the GPL-2 or later.
*/
#ifndef __AD1836_H__
@@ -21,39 +12,30 @@
#define AD1836_DAC_CTRL1 0
#define AD1836_DAC_POWERDOWN 2
-#define AD1836_DAC_SERFMT_MASK 0xE0
+#define AD1836_DAC_SERFMT_MASK 0xE0
#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5)
#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5)
#define AD1836_DAC_WORD_LEN_MASK 0x18
#define AD1836_DAC_WORD_LEN_OFFSET 3
#define AD1836_DAC_CTRL2 1
-#define AD1836_DACL1_MUTE 0
-#define AD1836_DACR1_MUTE 1
-#define AD1836_DACL2_MUTE 2
-#define AD1836_DACR2_MUTE 3
-#define AD1836_DACL3_MUTE 4
-#define AD1836_DACR3_MUTE 5
-#define AD1836_DAC_L1_VOL 2
-#define AD1836_DAC_R1_VOL 3
-#define AD1836_DAC_L2_VOL 4
-#define AD1836_DAC_R2_VOL 5
-#define AD1836_DAC_L3_VOL 6
-#define AD1836_DAC_R3_VOL 7
+/* These macros are one-based. So AD183X_MUTE_LEFT(1) will return the mute bit
+ * for the first ADC/DAC */
+#define AD1836_MUTE_LEFT(x) (((x) * 2) - 2)
+#define AD1836_MUTE_RIGHT(x) (((x) * 2) - 1)
+
+#define AD1836_DAC_L_VOL(x) ((x) * 2)
+#define AD1836_DAC_R_VOL(x) (1 + ((x) * 2))
#define AD1836_ADC_CTRL1 12
#define AD1836_ADC_POWERDOWN 7
#define AD1836_ADC_HIGHPASS_FILTER 8
#define AD1836_ADC_CTRL2 13
-#define AD1836_ADCL1_MUTE 0
-#define AD1836_ADCR1_MUTE 1
-#define AD1836_ADCL2_MUTE 2
-#define AD1836_ADCR2_MUTE 3
#define AD1836_ADC_WORD_LEN_MASK 0x30
#define AD1836_ADC_WORD_OFFSET 5
-#define AD1836_ADC_SERFMT_MASK (7 << 6)
+#define AD1836_ADC_SERFMT_MASK (7 << 6)
#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6)
#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6)
#define AD1836_ADC_AUX (0x6 << 6)
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
new file mode 100644
index 0000000..2758d5f
--- /dev/null
+++ b/sound/soc/codecs/adau1701.c
@@ -0,0 +1,549 @@
+/*
+ * Driver for ADAU1701 SigmaDSP processor
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ * based on an inital version by Cliff Cai <cliff.cai@analog.com>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <linux/sigma.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "adau1701.h"
+
+#define ADAU1701_DSPCTRL 0x1c
+#define ADAU1701_SEROCTL 0x1e
+#define ADAU1701_SERICTL 0x1f
+
+#define ADAU1701_AUXNPOW 0x22
+
+#define ADAU1701_OSCIPOW 0x26
+#define ADAU1701_DACSET 0x27
+
+#define ADAU1701_NUM_REGS 0x28
+
+#define ADAU1701_DSPCTRL_CR (1 << 2)
+#define ADAU1701_DSPCTRL_DAM (1 << 3)
+#define ADAU1701_DSPCTRL_ADM (1 << 4)
+#define ADAU1701_DSPCTRL_SR_48 0x00
+#define ADAU1701_DSPCTRL_SR_96 0x01
+#define ADAU1701_DSPCTRL_SR_192 0x02
+#define ADAU1701_DSPCTRL_SR_MASK 0x03
+
+#define ADAU1701_SEROCTL_INV_LRCLK 0x2000
+#define ADAU1701_SEROCTL_INV_BCLK 0x1000
+#define ADAU1701_SEROCTL_MASTER 0x0800
+
+#define ADAU1701_SEROCTL_OBF16 0x0000
+#define ADAU1701_SEROCTL_OBF8 0x0200
+#define ADAU1701_SEROCTL_OBF4 0x0400
+#define ADAU1701_SEROCTL_OBF2 0x0600
+#define ADAU1701_SEROCTL_OBF_MASK 0x0600
+
+#define ADAU1701_SEROCTL_OLF1024 0x0000
+#define ADAU1701_SEROCTL_OLF512 0x0080
+#define ADAU1701_SEROCTL_OLF256 0x0100
+#define ADAU1701_SEROCTL_OLF_MASK 0x0180
+
+#define ADAU1701_SEROCTL_MSB_DEALY1 0x0000
+#define ADAU1701_SEROCTL_MSB_DEALY0 0x0004
+#define ADAU1701_SEROCTL_MSB_DEALY8 0x0008
+#define ADAU1701_SEROCTL_MSB_DEALY12 0x000c
+#define ADAU1701_SEROCTL_MSB_DEALY16 0x0010
+#define ADAU1701_SEROCTL_MSB_DEALY_MASK 0x001c
+
+#define ADAU1701_SEROCTL_WORD_LEN_24 0x0000
+#define ADAU1701_SEROCTL_WORD_LEN_20 0x0001
+#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010
+#define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003
+
+#define ADAU1701_AUXNPOW_VBPD 0x40
+#define ADAU1701_AUXNPOW_VRPD 0x20
+
+#define ADAU1701_SERICTL_I2S 0
+#define ADAU1701_SERICTL_LEFTJ 1
+#define ADAU1701_SERICTL_TDM 2
+#define ADAU1701_SERICTL_RIGHTJ_24 3
+#define ADAU1701_SERICTL_RIGHTJ_20 4
+#define ADAU1701_SERICTL_RIGHTJ_18 5
+#define ADAU1701_SERICTL_RIGHTJ_16 6
+#define ADAU1701_SERICTL_MODE_MASK 7
+#define ADAU1701_SERICTL_INV_BCLK BIT(3)
+#define ADAU1701_SERICTL_INV_LRCLK BIT(4)
+
+#define ADAU1701_OSCIPOW_OPD 0x04
+#define ADAU1701_DACSET_DACINIT 1
+
+#define ADAU1701_FIRMWARE "adau1701.bin"
+
+struct adau1701 {
+ unsigned int dai_fmt;
+};
+
+static const struct snd_kcontrol_new adau1701_controls[] = {
+ SOC_SINGLE("Master Capture Switch", ADAU1701_DSPCTRL, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget adau1701_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC0", "Playback", ADAU1701_AUXNPOW, 3, 1),
+ SND_SOC_DAPM_DAC("DAC1", "Playback", ADAU1701_AUXNPOW, 2, 1),
+ SND_SOC_DAPM_DAC("DAC2", "Playback", ADAU1701_AUXNPOW, 1, 1),
+ SND_SOC_DAPM_DAC("DAC3", "Playback", ADAU1701_AUXNPOW, 0, 1),
+ SND_SOC_DAPM_ADC("ADC", "Capture", ADAU1701_AUXNPOW, 7, 1),
+
+ SND_SOC_DAPM_OUTPUT("OUT0"),
+ SND_SOC_DAPM_OUTPUT("OUT1"),
+ SND_SOC_DAPM_OUTPUT("OUT2"),
+ SND_SOC_DAPM_OUTPUT("OUT3"),
+ SND_SOC_DAPM_INPUT("IN0"),
+ SND_SOC_DAPM_INPUT("IN1"),
+};
+
+static const struct snd_soc_dapm_route adau1701_dapm_routes[] = {
+ { "OUT0", NULL, "DAC0" },
+ { "OUT1", NULL, "DAC1" },
+ { "OUT2", NULL, "DAC2" },
+ { "OUT3", NULL, "DAC3" },
+
+ { "ADC", NULL, "IN0" },
+ { "ADC", NULL, "IN1" },
+};
+
+static unsigned int adau1701_register_size(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ switch (reg) {
+ case ADAU1701_DSPCTRL:
+ case ADAU1701_SEROCTL:
+ case ADAU1701_AUXNPOW:
+ case ADAU1701_OSCIPOW:
+ case ADAU1701_DACSET:
+ return 2;
+ case ADAU1701_SERICTL:
+ return 1;
+ }
+
+ dev_err(codec->dev, "Unsupported register address: %d\n", reg);
+ return 0;
+}
+
+static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ unsigned int i;
+ unsigned int size;
+ uint8_t buf[4];
+ int ret;
+
+ size = adau1701_register_size(codec, reg);
+ if (size == 0)
+ return -EINVAL;
+
+ snd_soc_cache_write(codec, reg, value);
+
+ buf[0] = 0x08;
+ buf[1] = reg;
+
+ for (i = size + 1; i >= 2; --i) {
+ buf[i] = value;
+ value >>= 8;
+ }
+
+ ret = i2c_master_send(to_i2c_client(codec->dev), buf, size + 2);
+ if (ret == size + 2)
+ return 0;
+ else if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+
+static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ unsigned int value;
+ unsigned int ret;
+
+ ret = snd_soc_cache_read(codec, reg, &value);
+ if (ret)
+ return ret;
+
+ return value;
+}
+
+static int adau1701_load_firmware(struct snd_soc_codec *codec)
+{
+ return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE);
+}
+
+static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec,
+ snd_pcm_format_t format)
+{
+ struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+ unsigned int mask = ADAU1701_SEROCTL_WORD_LEN_MASK;
+ unsigned int val;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val = ADAU1701_SEROCTL_WORD_LEN_16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val = ADAU1701_SEROCTL_WORD_LEN_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val = ADAU1701_SEROCTL_WORD_LEN_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (adau1701->dai_fmt == SND_SOC_DAIFMT_RIGHT_J) {
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val |= ADAU1701_SEROCTL_MSB_DEALY16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val |= ADAU1701_SEROCTL_MSB_DEALY12;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val |= ADAU1701_SEROCTL_MSB_DEALY8;
+ break;
+ }
+ mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK;
+ }
+
+ snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val);
+
+ return 0;
+}
+
+static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec,
+ snd_pcm_format_t format)
+{
+ struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val;
+
+ if (adau1701->dai_fmt != SND_SOC_DAIFMT_RIGHT_J)
+ return 0;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val = ADAU1701_SERICTL_RIGHTJ_16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val = ADAU1701_SERICTL_RIGHTJ_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val = ADAU1701_SERICTL_RIGHTJ_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ADAU1701_SERICTL,
+ ADAU1701_SERICTL_MODE_MASK, val);
+
+ return 0;
+}
+
+static int adau1701_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ snd_pcm_format_t format;
+ unsigned int val;
+
+ switch (params_rate(params)) {
+ case 192000:
+ val = ADAU1701_DSPCTRL_SR_192;
+ break;
+ case 96000:
+ val = ADAU1701_DSPCTRL_SR_96;
+ break;
+ case 48000:
+ val = ADAU1701_DSPCTRL_SR_48;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ADAU1701_DSPCTRL,
+ ADAU1701_DSPCTRL_SR_MASK, val);
+
+ format = params_format(params);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return adau1701_set_playback_pcm_format(codec, format);
+ else
+ return adau1701_set_capture_pcm_format(codec, format);
+}
+
+static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+ unsigned int serictl = 0x00, seroctl = 0x00;
+ bool invert_lrclk;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ /* master, 64-bits per sample, 1 frame per sample */
+ seroctl |= ADAU1701_SEROCTL_MASTER | ADAU1701_SEROCTL_OBF16
+ | ADAU1701_SEROCTL_OLF1024;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ invert_lrclk = false;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ invert_lrclk = true;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ invert_lrclk = false;
+ serictl |= ADAU1701_SERICTL_INV_BCLK;
+ seroctl |= ADAU1701_SEROCTL_INV_BCLK;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ invert_lrclk = true;
+ serictl |= ADAU1701_SERICTL_INV_BCLK;
+ seroctl |= ADAU1701_SEROCTL_INV_BCLK;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ serictl |= ADAU1701_SERICTL_LEFTJ;
+ seroctl |= ADAU1701_SEROCTL_MSB_DEALY0;
+ invert_lrclk = !invert_lrclk;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ serictl |= ADAU1701_SERICTL_RIGHTJ_24;
+ seroctl |= ADAU1701_SEROCTL_MSB_DEALY8;
+ invert_lrclk = !invert_lrclk;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (invert_lrclk) {
+ seroctl |= ADAU1701_SEROCTL_INV_LRCLK;
+ serictl |= ADAU1701_SERICTL_INV_LRCLK;
+ }
+
+ adau1701->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+ snd_soc_write(codec, ADAU1701_SERICTL, serictl);
+ snd_soc_update_bits(codec, ADAU1701_SEROCTL,
+ ~ADAU1701_SEROCTL_WORD_LEN_MASK, seroctl);
+
+ return 0;
+}
+
+static int adau1701_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* Enable VREF and VREF buffer */
+ snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* Disable VREF and VREF buffer */
+ snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int mask = ADAU1701_DSPCTRL_DAM;
+ unsigned int val;
+
+ if (mute)
+ val = 0;
+ else
+ val = mask;
+
+ snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val);
+
+ return 0;
+}
+
+static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+ unsigned int freq, int dir)
+{
+ unsigned int val;
+
+ switch (clk_id) {
+ case ADAU1701_CLK_SRC_OSC:
+ val = 0x0;
+ break;
+ case ADAU1701_CLK_SRC_MCLK:
+ val = ADAU1701_OSCIPOW_OPD;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val);
+
+ return 0;
+}
+
+#define ADAU1701_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_192000)
+
+#define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static const struct snd_soc_dai_ops adau1701_dai_ops = {
+ .set_fmt = adau1701_set_dai_fmt,
+ .hw_params = adau1701_hw_params,
+ .digital_mute = adau1701_digital_mute,
+};
+
+static struct snd_soc_dai_driver adau1701_dai = {
+ .name = "adau1701",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = ADAU1701_RATES,
+ .formats = ADAU1701_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = ADAU1701_RATES,
+ .formats = ADAU1701_FORMATS,
+ },
+ .ops = &adau1701_dai_ops,
+ .symmetric_rates = 1,
+};
+
+static int adau1701_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ codec->dapm.idle_bias_off = 1;
+
+ ret = adau1701_load_firmware(codec);
+ if (ret)
+ dev_warn(codec->dev, "Failed to load firmware\n");
+
+ snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT);
+ snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver adau1701_codec_drv = {
+ .probe = adau1701_probe,
+ .set_bias_level = adau1701_set_bias_level,
+
+ .reg_cache_size = ADAU1701_NUM_REGS,
+ .reg_word_size = sizeof(u16),
+
+ .controls = adau1701_controls,
+ .num_controls = ARRAY_SIZE(adau1701_controls),
+ .dapm_widgets = adau1701_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(adau1701_dapm_widgets),
+ .dapm_routes = adau1701_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(adau1701_dapm_routes),
+
+ .write = adau1701_write,
+ .read = adau1701_read,
+
+ .set_sysclk = adau1701_set_sysclk,
+};
+
+static __devinit int adau1701_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct adau1701 *adau1701;
+ int ret;
+
+ adau1701 = kzalloc(sizeof(*adau1701), GFP_KERNEL);
+ if (!adau1701)
+ return -ENOMEM;
+
+ i2c_set_clientdata(client, adau1701);
+ ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv,
+ &adau1701_dai, 1);
+ if (ret < 0)
+ kfree(adau1701);
+
+ return ret;
+}
+
+static __devexit int adau1701_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ kfree(i2c_get_clientdata(client));
+ return 0;
+}
+
+static const struct i2c_device_id adau1701_i2c_id[] = {
+ { "adau1701", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id);
+
+static struct i2c_driver adau1701_i2c_driver = {
+ .driver = {
+ .name = "adau1701",
+ .owner = THIS_MODULE,
+ },
+ .probe = adau1701_i2c_probe,
+ .remove = __devexit_p(adau1701_i2c_remove),
+ .id_table = adau1701_i2c_id,
+};
+
+static int __init adau1701_init(void)
+{
+ return i2c_add_driver(&adau1701_i2c_driver);
+}
+module_init(adau1701_init);
+
+static void __exit adau1701_exit(void)
+{
+ i2c_del_driver(&adau1701_i2c_driver);
+}
+module_exit(adau1701_exit);
+
+MODULE_DESCRIPTION("ASoC ADAU1701 SigmaDSP driver");
+MODULE_AUTHOR("Cliff Cai <cliff.cai@analog.com>");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1701.h b/sound/soc/codecs/adau1701.h
new file mode 100644
index 0000000..8d0949a
--- /dev/null
+++ b/sound/soc/codecs/adau1701.h
@@ -0,0 +1,17 @@
+/*
+ * header file for ADAU1701 SigmaDSP processor
+ *
+ * Copyright 2011 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef _ADAU1701_H
+#define _ADAU1701_H
+
+enum adau1701_clk_src {
+ ADAU1701_CLK_SRC_OSC,
+ ADAU1701_CLK_SRC_MCLK,
+};
+
+#endif
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
new file mode 100644
index 0000000..e30fba6
--- /dev/null
+++ b/sound/soc/codecs/adav80x.c
@@ -0,0 +1,951 @@
+/*
+ * ADAV80X Audio Codec driver supporting ADAV801, ADAV803
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Yi Li <yi.li@analog.com>
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+
+#include "adav80x.h"
+
+#define ADAV80X_PLAYBACK_CTRL 0x04
+#define ADAV80X_AUX_IN_CTRL 0x05
+#define ADAV80X_REC_CTRL 0x06
+#define ADAV80X_AUX_OUT_CTRL 0x07
+#define ADAV80X_DPATH_CTRL1 0x62
+#define ADAV80X_DPATH_CTRL2 0x63
+#define ADAV80X_DAC_CTRL1 0x64
+#define ADAV80X_DAC_CTRL2 0x65
+#define ADAV80X_DAC_CTRL3 0x66
+#define ADAV80X_DAC_L_VOL 0x68
+#define ADAV80X_DAC_R_VOL 0x69
+#define ADAV80X_PGA_L_VOL 0x6c
+#define ADAV80X_PGA_R_VOL 0x6d
+#define ADAV80X_ADC_CTRL1 0x6e
+#define ADAV80X_ADC_CTRL2 0x6f
+#define ADAV80X_ADC_L_VOL 0x70
+#define ADAV80X_ADC_R_VOL 0x71
+#define ADAV80X_PLL_CTRL1 0x74
+#define ADAV80X_PLL_CTRL2 0x75
+#define ADAV80X_ICLK_CTRL1 0x76
+#define ADAV80X_ICLK_CTRL2 0x77
+#define ADAV80X_PLL_CLK_SRC 0x78
+#define ADAV80X_PLL_OUTE 0x7a
+
+#define ADAV80X_PLL_CLK_SRC_PLL_XIN(pll) 0x00
+#define ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll) (0x40 << (pll))
+#define ADAV80X_PLL_CLK_SRC_PLL_MASK(pll) (0x40 << (pll))
+
+#define ADAV80X_ICLK_CTRL1_DAC_SRC(src) ((src) << 5)
+#define ADAV80X_ICLK_CTRL1_ADC_SRC(src) ((src) << 2)
+#define ADAV80X_ICLK_CTRL1_ICLK2_SRC(src) (src)
+#define ADAV80X_ICLK_CTRL2_ICLK1_SRC(src) ((src) << 3)
+
+#define ADAV80X_PLL_CTRL1_PLLDIV 0x10
+#define ADAV80X_PLL_CTRL1_PLLPD(pll) (0x04 << (pll))
+#define ADAV80X_PLL_CTRL1_XTLPD 0x02
+
+#define ADAV80X_PLL_CTRL2_FIELD(pll, x) ((x) << ((pll) * 4))
+
+#define ADAV80X_PLL_CTRL2_FS_48(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x00)
+#define ADAV80X_PLL_CTRL2_FS_32(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x08)
+#define ADAV80X_PLL_CTRL2_FS_44(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0c)
+
+#define ADAV80X_PLL_CTRL2_SEL(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x02)
+#define ADAV80X_PLL_CTRL2_DOUB(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x01)
+#define ADAV80X_PLL_CTRL2_PLL_MASK(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0f)
+
+#define ADAV80X_ADC_CTRL1_MODULATOR_MASK 0x80
+#define ADAV80X_ADC_CTRL1_MODULATOR_128FS 0x00
+#define ADAV80X_ADC_CTRL1_MODULATOR_64FS 0x80
+
+#define ADAV80X_DAC_CTRL1_PD 0x80
+
+#define ADAV80X_DAC_CTRL2_DIV1 0x00
+#define ADAV80X_DAC_CTRL2_DIV1_5 0x10
+#define ADAV80X_DAC_CTRL2_DIV2 0x20
+#define ADAV80X_DAC_CTRL2_DIV3 0x30
+#define ADAV80X_DAC_CTRL2_DIV_MASK 0x30
+
+#define ADAV80X_DAC_CTRL2_INTERPOL_256FS 0x00
+#define ADAV80X_DAC_CTRL2_INTERPOL_128FS 0x40
+#define ADAV80X_DAC_CTRL2_INTERPOL_64FS 0x80
+#define ADAV80X_DAC_CTRL2_INTERPOL_MASK 0xc0
+
+#define ADAV80X_DAC_CTRL2_DEEMPH_NONE 0x00
+#define ADAV80X_DAC_CTRL2_DEEMPH_44 0x01
+#define ADAV80X_DAC_CTRL2_DEEMPH_32 0x02
+#define ADAV80X_DAC_CTRL2_DEEMPH_48 0x03
+#define ADAV80X_DAC_CTRL2_DEEMPH_MASK 0x01
+
+#define ADAV80X_CAPTURE_MODE_MASTER 0x20
+#define ADAV80X_CAPTURE_WORD_LEN24 0x00
+#define ADAV80X_CAPTURE_WORD_LEN20 0x04
+#define ADAV80X_CAPTRUE_WORD_LEN18 0x08
+#define ADAV80X_CAPTURE_WORD_LEN16 0x0c
+#define ADAV80X_CAPTURE_WORD_LEN_MASK 0x0c
+
+#define ADAV80X_CAPTURE_MODE_LEFT_J 0x00
+#define ADAV80X_CAPTURE_MODE_I2S 0x01
+#define ADAV80X_CAPTURE_MODE_RIGHT_J 0x03
+#define ADAV80X_CAPTURE_MODE_MASK 0x03
+
+#define ADAV80X_PLAYBACK_MODE_MASTER 0x10
+#define ADAV80X_PLAYBACK_MODE_LEFT_J 0x00
+#define ADAV80X_PLAYBACK_MODE_I2S 0x01
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_24 0x04
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_20 0x05
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_18 0x06
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_16 0x07
+#define ADAV80X_PLAYBACK_MODE_MASK 0x07
+
+#define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x))
+
+static u8 adav80x_default_regs[] = {
+ 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00,
+ 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37,
+ 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b,
+ 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00,
+ 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee,
+ 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f,
+ 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x52, 0x00,
+};
+
+struct adav80x {
+ enum snd_soc_control_type control_type;
+
+ enum adav80x_clk_src clk_src;
+ unsigned int sysclk;
+ enum adav80x_pll_src pll_src;
+
+ unsigned int dai_fmt[2];
+ unsigned int rate;
+ bool deemph;
+ bool sysclk_pd[3];
+};
+
+static const char *adav80x_mux_text[] = {
+ "ADC",
+ "Playback",
+ "Aux Playback",
+};
+
+static const unsigned int adav80x_mux_values[] = {
+ 0, 2, 3,
+};
+
+#define ADAV80X_MUX_ENUM_DECL(name, reg, shift) \
+ SOC_VALUE_ENUM_DOUBLE_DECL(name, reg, shift, 7, \
+ ARRAY_SIZE(adav80x_mux_text), adav80x_mux_text, \
+ adav80x_mux_values)
+
+static ADAV80X_MUX_ENUM_DECL(adav80x_aux_capture_enum, ADAV80X_DPATH_CTRL1, 0);
+static ADAV80X_MUX_ENUM_DECL(adav80x_capture_enum, ADAV80X_DPATH_CTRL1, 3);
+static ADAV80X_MUX_ENUM_DECL(adav80x_dac_enum, ADAV80X_DPATH_CTRL2, 3);
+
+static const struct snd_kcontrol_new adav80x_aux_capture_mux_ctrl =
+ SOC_DAPM_VALUE_ENUM("Route", adav80x_aux_capture_enum);
+static const struct snd_kcontrol_new adav80x_capture_mux_ctrl =
+ SOC_DAPM_VALUE_ENUM("Route", adav80x_capture_enum);
+static const struct snd_kcontrol_new adav80x_dac_mux_ctrl =
+ SOC_DAPM_VALUE_ENUM("Route", adav80x_dac_enum);
+
+#define ADAV80X_MUX(name, ctrl) \
+ SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl)
+
+static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", NULL, ADAV80X_DAC_CTRL1, 7, 1),
+ SND_SOC_DAPM_ADC("ADC", NULL, ADAV80X_ADC_CTRL1, 5, 1),
+
+ SND_SOC_DAPM_PGA("Right PGA", ADAV80X_ADC_CTRL1, 0, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("Left PGA", ADAV80X_ADC_CTRL1, 1, 1, NULL, 0),
+
+ SND_SOC_DAPM_AIF_OUT("AIFOUT", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIFIN", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_AIF_OUT("AIFAUXOUT", "Aux Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIFAUXIN", "Aux Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ ADAV80X_MUX("Aux Capture Select", &adav80x_aux_capture_mux_ctrl),
+ ADAV80X_MUX("Capture Select", &adav80x_capture_mux_ctrl),
+ ADAV80X_MUX("DAC Select", &adav80x_dac_mux_ctrl),
+
+ SND_SOC_DAPM_INPUT("VINR"),
+ SND_SOC_DAPM_INPUT("VINL"),
+ SND_SOC_DAPM_OUTPUT("VOUTR"),
+ SND_SOC_DAPM_OUTPUT("VOUTL"),
+
+ SND_SOC_DAPM_SUPPLY("SYSCLK", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL1", ADAV80X_PLL_CTRL1, 2, 1, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL2", ADAV80X_PLL_CTRL1, 3, 1, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("OSC", ADAV80X_PLL_CTRL1, 1, 1, NULL, 0),
+};
+
+static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct snd_soc_codec *codec = source->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ const char *clk;
+
+ switch (adav80x->clk_src) {
+ case ADAV80X_CLK_PLL1:
+ clk = "PLL1";
+ break;
+ case ADAV80X_CLK_PLL2:
+ clk = "PLL2";
+ break;
+ case ADAV80X_CLK_XTAL:
+ clk = "OSC";
+ break;
+ default:
+ return 0;
+ }
+
+ return strcmp(source->name, clk) == 0;
+}
+
+static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct snd_soc_codec *codec = source->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL;
+}
+
+
+static const struct snd_soc_dapm_route adav80x_dapm_routes[] = {
+ { "DAC Select", "ADC", "ADC" },
+ { "DAC Select", "Playback", "AIFIN" },
+ { "DAC Select", "Aux Playback", "AIFAUXIN" },
+ { "DAC", NULL, "DAC Select" },
+
+ { "Capture Select", "ADC", "ADC" },
+ { "Capture Select", "Playback", "AIFIN" },
+ { "Capture Select", "Aux Playback", "AIFAUXIN" },
+ { "AIFOUT", NULL, "Capture Select" },
+
+ { "Aux Capture Select", "ADC", "ADC" },
+ { "Aux Capture Select", "Playback", "AIFIN" },
+ { "Aux Capture Select", "Aux Playback", "AIFAUXIN" },
+ { "AIFAUXOUT", NULL, "Aux Capture Select" },
+
+ { "VOUTR", NULL, "DAC" },
+ { "VOUTL", NULL, "DAC" },
+
+ { "Left PGA", NULL, "VINL" },
+ { "Right PGA", NULL, "VINR" },
+ { "ADC", NULL, "Left PGA" },
+ { "ADC", NULL, "Right PGA" },
+
+ { "SYSCLK", NULL, "PLL1", adav80x_dapm_sysclk_check },
+ { "SYSCLK", NULL, "PLL2", adav80x_dapm_sysclk_check },
+ { "SYSCLK", NULL, "OSC", adav80x_dapm_sysclk_check },
+ { "PLL1", NULL, "OSC", adav80x_dapm_pll_check },
+ { "PLL2", NULL, "OSC", adav80x_dapm_pll_check },
+
+ { "ADC", NULL, "SYSCLK" },
+ { "DAC", NULL, "SYSCLK" },
+ { "AIFOUT", NULL, "SYSCLK" },
+ { "AIFAUXOUT", NULL, "SYSCLK" },
+ { "AIFIN", NULL, "SYSCLK" },
+ { "AIFAUXIN", NULL, "SYSCLK" },
+};
+
+static int adav80x_set_deemph(struct snd_soc_codec *codec)
+{
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int val;
+
+ if (adav80x->deemph) {
+ switch (adav80x->rate) {
+ case 32000:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_32;
+ break;
+ case 44100:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_44;
+ break;
+ case 48000:
+ case 64000:
+ case 88200:
+ case 96000:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_48;
+ break;
+ default:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
+ break;
+ }
+ } else {
+ val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
+ }
+
+ return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+ ADAV80X_DAC_CTRL2_DEEMPH_MASK, val);
+}
+
+static int adav80x_put_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int deemph = ucontrol->value.enumerated.item[0];
+
+ if (deemph > 1)
+ return -EINVAL;
+
+ adav80x->deemph = deemph;
+
+ return adav80x_set_deemph(codec);
+}
+
+static int adav80x_get_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.enumerated.item[0] = adav80x->deemph;
+ return 0;
+};
+
+static const DECLARE_TLV_DB_SCALE(adav80x_inpga_tlv, 0, 50, 0);
+static const DECLARE_TLV_DB_MINMAX(adav80x_digital_tlv, -9563, 0);
+
+static const struct snd_kcontrol_new adav80x_controls[] = {
+ SOC_DOUBLE_R_TLV("Master Playback Volume", ADAV80X_DAC_L_VOL,
+ ADAV80X_DAC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv),
+ SOC_DOUBLE_R_TLV("Master Capture Volume", ADAV80X_ADC_L_VOL,
+ ADAV80X_ADC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv),
+
+ SOC_DOUBLE_R_TLV("PGA Capture Volume", ADAV80X_PGA_L_VOL,
+ ADAV80X_PGA_R_VOL, 0, 0x30, 0, adav80x_inpga_tlv),
+
+ SOC_DOUBLE("Master Playback Switch", ADAV80X_DAC_CTRL1, 0, 1, 1, 0),
+ SOC_DOUBLE("Master Capture Switch", ADAV80X_ADC_CTRL1, 2, 3, 1, 1),
+
+ SOC_SINGLE("ADC High Pass Filter Switch", ADAV80X_ADC_CTRL1, 6, 1, 0),
+
+ SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0,
+ adav80x_get_deemph, adav80x_put_deemph),
+};
+
+static unsigned int adav80x_port_ctrl_regs[2][2] = {
+ { ADAV80X_REC_CTRL, ADAV80X_PLAYBACK_CTRL, },
+ { ADAV80X_AUX_OUT_CTRL, ADAV80X_AUX_IN_CTRL },
+};
+
+static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int capture = 0x00;
+ unsigned int playback = 0x00;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ capture |= ADAV80X_CAPTURE_MODE_MASTER;
+ playback |= ADAV80X_PLAYBACK_MODE_MASTER;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ capture |= ADAV80X_CAPTURE_MODE_I2S;
+ playback |= ADAV80X_PLAYBACK_MODE_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ capture |= ADAV80X_CAPTURE_MODE_LEFT_J;
+ playback |= ADAV80X_PLAYBACK_MODE_LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ capture |= ADAV80X_CAPTURE_MODE_RIGHT_J;
+ playback |= ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+ ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER,
+ capture);
+ snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback);
+
+ adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+ return 0;
+}
+
+static int adav80x_set_adc_clock(struct snd_soc_codec *codec,
+ unsigned int sample_rate)
+{
+ unsigned int val;
+
+ if (sample_rate <= 48000)
+ val = ADAV80X_ADC_CTRL1_MODULATOR_128FS;
+ else
+ val = ADAV80X_ADC_CTRL1_MODULATOR_64FS;
+
+ snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1,
+ ADAV80X_ADC_CTRL1_MODULATOR_MASK, val);
+
+ return 0;
+}
+
+static int adav80x_set_dac_clock(struct snd_soc_codec *codec,
+ unsigned int sample_rate)
+{
+ unsigned int val;
+
+ if (sample_rate <= 48000)
+ val = ADAV80X_DAC_CTRL2_DIV1 | ADAV80X_DAC_CTRL2_INTERPOL_256FS;
+ else
+ val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS;
+
+ snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+ ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK,
+ val);
+
+ return 0;
+}
+
+static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec,
+ struct snd_soc_dai *dai, snd_pcm_format_t format)
+{
+ unsigned int val;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val = ADAV80X_CAPTURE_WORD_LEN16;
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ val = ADAV80X_CAPTRUE_WORD_LEN18;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val = ADAV80X_CAPTURE_WORD_LEN20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val = ADAV80X_CAPTURE_WORD_LEN24;
+ break;
+ default:
+ break;
+ }
+
+ snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+ ADAV80X_CAPTURE_WORD_LEN_MASK, val);
+
+ return 0;
+}
+
+static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec,
+ struct snd_soc_dai *dai, snd_pcm_format_t format)
+{
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int val;
+
+ if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J)
+ return 0;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16;
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
+ break;
+ default:
+ break;
+ }
+
+ snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1],
+ ADAV80X_PLAYBACK_MODE_MASK, val);
+
+ return 0;
+}
+
+static int adav80x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int rate = params_rate(params);
+
+ if (rate * 256 != adav80x->sysclk)
+ return -EINVAL;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ adav80x_set_playback_pcm_format(codec, dai,
+ params_format(params));
+ adav80x_set_dac_clock(codec, rate);
+ } else {
+ adav80x_set_capture_pcm_format(codec, dai,
+ params_format(params));
+ adav80x_set_adc_clock(codec, rate);
+ }
+ adav80x->rate = rate;
+ adav80x_set_deemph(codec);
+
+ return 0;
+}
+
+static int adav80x_set_sysclk(struct snd_soc_codec *codec,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ if (dir == SND_SOC_CLOCK_IN) {
+ switch (clk_id) {
+ case ADAV80X_CLK_XIN:
+ case ADAV80X_CLK_XTAL:
+ case ADAV80X_CLK_MCLKI:
+ case ADAV80X_CLK_PLL1:
+ case ADAV80X_CLK_PLL2:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ adav80x->sysclk = freq;
+
+ if (adav80x->clk_src != clk_id) {
+ unsigned int iclk_ctrl1, iclk_ctrl2;
+
+ adav80x->clk_src = clk_id;
+ if (clk_id == ADAV80X_CLK_XTAL)
+ clk_id = ADAV80X_CLK_XIN;
+
+ iclk_ctrl1 = ADAV80X_ICLK_CTRL1_DAC_SRC(clk_id) |
+ ADAV80X_ICLK_CTRL1_ADC_SRC(clk_id) |
+ ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id);
+ iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id);
+
+ snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1);
+ snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2);
+
+ snd_soc_dapm_sync(&codec->dapm);
+ }
+ } else {
+ unsigned int mask;
+
+ switch (clk_id) {
+ case ADAV80X_CLK_SYSCLK1:
+ case ADAV80X_CLK_SYSCLK2:
+ case ADAV80X_CLK_SYSCLK3:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ clk_id -= ADAV80X_CLK_SYSCLK1;
+ mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id);
+
+ if (freq == 0) {
+ snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask);
+ adav80x->sysclk_pd[clk_id] = true;
+ } else {
+ snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0);
+ adav80x->sysclk_pd[clk_id] = false;
+ }
+
+ if (adav80x->sysclk_pd[0])
+ snd_soc_dapm_disable_pin(&codec->dapm, "PLL1");
+ else
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+
+ if (adav80x->sysclk_pd[1] || adav80x->sysclk_pd[2])
+ snd_soc_dapm_disable_pin(&codec->dapm, "PLL2");
+ else
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+
+ snd_soc_dapm_sync(&codec->dapm);
+ }
+
+ return 0;
+}
+
+static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
+{
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int pll_ctrl1 = 0;
+ unsigned int pll_ctrl2 = 0;
+ unsigned int pll_src;
+
+ switch (source) {
+ case ADAV80X_PLL_SRC_XTAL:
+ case ADAV80X_PLL_SRC_XIN:
+ case ADAV80X_PLL_SRC_MCLKI:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (!freq_out)
+ return 0;
+
+ switch (freq_in) {
+ case 27000000:
+ break;
+ case 54000000:
+ if (source == ADAV80X_PLL_SRC_XIN) {
+ pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV;
+ break;
+ }
+ default:
+ return -EINVAL;
+ }
+
+ if (freq_out > 12288000) {
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_DOUB(pll_id);
+ freq_out /= 2;
+ }
+
+ /* freq_out = sample_rate * 256 */
+ switch (freq_out) {
+ case 8192000:
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_32(pll_id);
+ break;
+ case 11289600:
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_44(pll_id);
+ break;
+ case 12288000:
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_48(pll_id);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV,
+ pll_ctrl1);
+ snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2,
+ ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2);
+
+ if (source != adav80x->pll_src) {
+ if (source == ADAV80X_PLL_SRC_MCLKI)
+ pll_src = ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll_id);
+ else
+ pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id);
+
+ snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC,
+ ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src);
+
+ adav80x->pll_src = source;
+
+ snd_soc_dapm_sync(&codec->dapm);
+ }
+
+ return 0;
+}
+
+static int adav80x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ unsigned int mask = ADAV80X_DAC_CTRL1_PD;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+/* Enforce the same sample rate on all audio interfaces */
+static int adav80x_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ if (!codec->active || !adav80x->rate)
+ return 0;
+
+ return snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate);
+}
+
+static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ if (!codec->active)
+ adav80x->rate = 0;
+}
+
+static const struct snd_soc_dai_ops adav80x_dai_ops = {
+ .set_fmt = adav80x_set_dai_fmt,
+ .hw_params = adav80x_hw_params,
+ .startup = adav80x_dai_startup,
+ .shutdown = adav80x_dai_shutdown,
+};
+
+#define ADAV80X_PLAYBACK_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000)
+
+#define ADAV80X_CAPTURE_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+#define ADAV80X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver adav80x_dais[] = {
+ {
+ .name = "adav80x-hifi",
+ .id = 0,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_PLAYBACK_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_CAPTURE_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .ops = &adav80x_dai_ops,
+ },
+ {
+ .name = "adav80x-aux",
+ .id = 1,
+ .playback = {
+ .stream_name = "Aux Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_PLAYBACK_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Aux Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_CAPTURE_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .ops = &adav80x_dai_ops,
+ },
+};
+
+static int adav80x_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type);
+ if (ret) {
+ dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ /* Force PLLs on for SYSCLK output */
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+
+ /* Power down S/PDIF receiver, since it is currently not supported */
+ snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20);
+ /* Disable DAC zero flag */
+ snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6);
+
+ return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+}
+
+static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int adav80x_resume(struct snd_soc_codec *codec)
+{
+ adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ codec->cache_sync = 1;
+ snd_soc_cache_sync(codec);
+
+ return 0;
+}
+
+static int adav80x_remove(struct snd_soc_codec *codec)
+{
+ return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static struct snd_soc_codec_driver adav80x_codec_driver = {
+ .probe = adav80x_probe,
+ .remove = adav80x_remove,
+ .suspend = adav80x_suspend,
+ .resume = adav80x_resume,
+ .set_bias_level = adav80x_set_bias_level,
+
+ .set_pll = adav80x_set_pll,
+ .set_sysclk = adav80x_set_sysclk,
+
+ .reg_word_size = sizeof(u8),
+ .reg_cache_size = ARRAY_SIZE(adav80x_default_regs),
+ .reg_cache_default = adav80x_default_regs,
+
+ .controls = adav80x_controls,
+ .num_controls = ARRAY_SIZE(adav80x_controls),
+ .dapm_widgets = adav80x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(adav80x_dapm_widgets),
+ .dapm_routes = adav80x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes),
+};
+
+static int __devinit adav80x_bus_probe(struct device *dev,
+ enum snd_soc_control_type control_type)
+{
+ struct adav80x *adav80x;
+ int ret;
+
+ adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL);
+ if (!adav80x)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, adav80x);
+ adav80x->control_type = control_type;
+
+ ret = snd_soc_register_codec(dev, &adav80x_codec_driver,
+ adav80x_dais, ARRAY_SIZE(adav80x_dais));
+ if (ret)
+ kfree(adav80x);
+
+ return ret;
+}
+
+static int __devexit adav80x_bus_remove(struct device *dev)
+{
+ snd_soc_unregister_codec(dev);
+ kfree(dev_get_drvdata(dev));
+ return 0;
+}
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit adav80x_spi_probe(struct spi_device *spi)
+{
+ return adav80x_bus_probe(&spi->dev, SND_SOC_SPI);
+}
+
+static int __devexit adav80x_spi_remove(struct spi_device *spi)
+{
+ return adav80x_bus_remove(&spi->dev);
+}
+
+static struct spi_driver adav80x_spi_driver = {
+ .driver = {
+ .name = "adav801",
+ .owner = THIS_MODULE,
+ },
+ .probe = adav80x_spi_probe,
+ .remove = __devexit_p(adav80x_spi_remove),
+};
+#endif
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static const struct i2c_device_id adav80x_id[] = {
+ { "adav803", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, adav80x_id);
+
+static int __devinit adav80x_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ return adav80x_bus_probe(&client->dev, SND_SOC_I2C);
+}
+
+static int __devexit adav80x_i2c_remove(struct i2c_client *client)
+{
+ return adav80x_bus_remove(&client->dev);
+}
+
+static struct i2c_driver adav80x_i2c_driver = {
+ .driver = {
+ .name = "adav803",
+ .owner = THIS_MODULE,
+ },
+ .probe = adav80x_i2c_probe,
+ .remove = __devexit_p(adav80x_i2c_remove),
+ .id_table = adav80x_id,
+};
+#endif
+
+static int __init adav80x_init(void)
+{
+ int ret = 0;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&adav80x_i2c_driver);
+ if (ret)
+ return ret;
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&adav80x_spi_driver);
+#endif
+
+ return ret;
+}
+module_init(adav80x_init);
+
+static void __exit adav80x_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&adav80x_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&adav80x_spi_driver);
+#endif
+}
+module_exit(adav80x_exit);
+
+MODULE_DESCRIPTION("ASoC ADAV80x driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_AUTHOR("Yi Li <yi.li@analog.com>>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h
new file mode 100644
index 0000000..adb0fc7
--- /dev/null
+++ b/sound/soc/codecs/adav80x.h
@@ -0,0 +1,35 @@
+/*
+ * header file for ADAV80X parts
+ *
+ * Copyright 2011 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef _ADAV80X_H
+#define _ADAV80X_H
+
+enum adav80x_pll_src {
+ ADAV80X_PLL_SRC_XIN,
+ ADAV80X_PLL_SRC_XTAL,
+ ADAV80X_PLL_SRC_MCLKI,
+};
+
+enum adav80x_pll {
+ ADAV80X_PLL1 = 0,
+ ADAV80X_PLL2 = 1,
+};
+
+enum adav80x_clk_src {
+ ADAV80X_CLK_XIN = 0,
+ ADAV80X_CLK_MCLKI = 1,
+ ADAV80X_CLK_PLL1 = 2,
+ ADAV80X_CLK_PLL2 = 3,
+ ADAV80X_CLK_XTAL = 6,
+
+ ADAV80X_CLK_SYSCLK1 = 6,
+ ADAV80X_CLK_SYSCLK2 = 7,
+ ADAV80X_CLK_SYSCLK3 = 8,
+};
+
+#endif
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index ed96f247c..7a64e58 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -457,7 +457,7 @@
.set_sysclk = ak4641_set_dai_sysclk,
};
-struct snd_soc_dai_driver ak4641_dai[] = {
+static struct snd_soc_dai_driver ak4641_dai[] = {
{
.name = "ak4641-hifi",
.id = 1,
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 0206a17..6cc8678 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -636,10 +636,7 @@
#endif /* CONFIG_PM */
/*
- * ASoC codec device structure
- *
- * Assign this variable to the codec_dev field of the machine driver's
- * snd_soc_device structure.
+ * ASoC codec driver structure
*/
static const struct snd_soc_codec_driver soc_codec_device_cs4270 = {
.probe = cs4270_probe,
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 4173b67..ac65a2d 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1397,8 +1397,6 @@
if (freq == max98088->sysclk)
return 0;
- max98088->sysclk = freq; /* remember current sysclk */
-
/* Setup clocks for slave mode, and using the PLL
* PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
* 0x02 (when master clk is 20MHz to 30MHz)..
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index e1d282d..668434d 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1517,8 +1517,6 @@
if (freq == max98095->sysclk)
return 0;
- max98095->sysclk = freq; /* remember current sysclk */
-
/* Setup clocks for slave mode, and using the PLL
* PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
* 0x02 (when master clk is 20MHz to 40MHz)..
@@ -2261,11 +2259,11 @@
ret = snd_soc_read(codec, M98095_0FF_REV_ID);
if (ret < 0) {
- dev_err(codec->dev, "Failed to read device revision: %d\n",
+ dev_err(codec->dev, "Failure reading hardware revision: %d\n",
ret);
goto err_access;
}
- dev_info(codec->dev, "revision %c\n", ret + 'A');
+ dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A');
snd_soc_write(codec, M98095_097_PWR_SYS, M98095_PWRSV);
@@ -2342,8 +2340,8 @@
max98095->control_data = i2c;
max98095->pdata = i2c->dev.platform_data;
- ret = snd_soc_register_codec(&i2c->dev,
- &soc_codec_dev_max98095, &max98095_dai[0], 3);
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095,
+ max98095_dai, ARRAY_SIZE(max98095_dai));
if (ret < 0)
kfree(max98095);
return ret;
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
new file mode 100644
index 0000000..486628a
--- /dev/null
+++ b/sound/soc/codecs/sta32x.c
@@ -0,0 +1,777 @@
+/*
+ * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on code from:
+ * Wolfson Microelectronics PLC.
+ * Mark Brown <broonie@opensource.wolfsonmicro.com>
+ * Freescale Semiconductor, Inc.
+ * Timur Tabi <timur@freescale.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.
+ */
+
+#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "sta32x.h"
+
+#define STA32X_RATES (SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | \
+ SNDRV_PCM_RATE_192000)
+
+#define STA32X_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE)
+
+/* Power-up register defaults */
+static const u8 sta32x_regs[STA32X_REGISTER_COUNT] = {
+ 0x63, 0x80, 0xc2, 0x40, 0xc2, 0x5c, 0x10, 0xff, 0x60, 0x60,
+ 0x60, 0x80, 0x00, 0x00, 0x00, 0x40, 0x80, 0x77, 0x6a, 0x69,
+ 0x6a, 0x69, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2d,
+ 0xc0, 0xf3, 0x33, 0x00, 0x0c,
+};
+
+/* regulator power supply names */
+static const char *sta32x_supply_names[] = {
+ "Vdda", /* analog supply, 3.3VV */
+ "Vdd3", /* digital supply, 3.3V */
+ "Vcc" /* power amp spply, 10V - 36V */
+};
+
+/* codec private data */
+struct sta32x_priv {
+ struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)];
+ struct snd_soc_codec *codec;
+
+ unsigned int mclk;
+ unsigned int format;
+};
+
+static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1);
+static const DECLARE_TLV_DB_SCALE(tone_tlv, -120, 200, 0);
+
+static const char *sta32x_drc_ac[] = {
+ "Anti-Clipping", "Dynamic Range Compression" };
+static const char *sta32x_auto_eq_mode[] = {
+ "User", "Preset", "Loudness" };
+static const char *sta32x_auto_gc_mode[] = {
+ "User", "AC no clipping", "AC limited clipping (10%)",
+ "DRC nighttime listening mode" };
+static const char *sta32x_auto_xo_mode[] = {
+ "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", "200Hz",
+ "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", "340Hz", "360Hz" };
+static const char *sta32x_preset_eq_mode[] = {
+ "Flat", "Rock", "Soft Rock", "Jazz", "Classical", "Dance", "Pop", "Soft",
+ "Hard", "Party", "Vocal", "Hip-Hop", "Dialog", "Bass-boost #1",
+ "Bass-boost #2", "Bass-boost #3", "Loudness 1", "Loudness 2",
+ "Loudness 3", "Loudness 4", "Loudness 5", "Loudness 6", "Loudness 7",
+ "Loudness 8", "Loudness 9", "Loudness 10", "Loudness 11", "Loudness 12",
+ "Loudness 13", "Loudness 14", "Loudness 15", "Loudness 16" };
+static const char *sta32x_limiter_select[] = {
+ "Limiter Disabled", "Limiter #1", "Limiter #2" };
+static const char *sta32x_limiter_attack_rate[] = {
+ "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024",
+ "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752",
+ "0.0645", "0.0564", "0.0501", "0.0451" };
+static const char *sta32x_limiter_release_rate[] = {
+ "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299",
+ "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137",
+ "0.0134", "0.0117", "0.0110", "0.0104" };
+
+static const unsigned int sta32x_limiter_ac_attack_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0),
+ 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0),
+};
+
+static const unsigned int sta32x_limiter_ac_release_tlv[] = {
+ TLV_DB_RANGE_HEAD(5),
+ 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0),
+ 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0),
+ 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0),
+};
+
+static const unsigned int sta32x_limiter_drc_attack_tlv[] = {
+ TLV_DB_RANGE_HEAD(3),
+ 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0),
+ 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0),
+ 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0),
+};
+
+static const unsigned int sta32x_limiter_drc_release_tlv[] = {
+ TLV_DB_RANGE_HEAD(5),
+ 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0),
+ 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0),
+ 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0),
+ 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0),
+ 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0),
+};
+
+static const struct soc_enum sta32x_drc_ac_enum =
+ SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
+ 2, sta32x_drc_ac);
+static const struct soc_enum sta32x_auto_eq_enum =
+ SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
+ 3, sta32x_auto_eq_mode);
+static const struct soc_enum sta32x_auto_gc_enum =
+ SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
+ 4, sta32x_auto_gc_mode);
+static const struct soc_enum sta32x_auto_xo_enum =
+ SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
+ 16, sta32x_auto_xo_mode);
+static const struct soc_enum sta32x_preset_eq_enum =
+ SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
+ 32, sta32x_preset_eq_mode);
+static const struct soc_enum sta32x_limiter_ch1_enum =
+ SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
+ 3, sta32x_limiter_select);
+static const struct soc_enum sta32x_limiter_ch2_enum =
+ SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
+ 3, sta32x_limiter_select);
+static const struct soc_enum sta32x_limiter_ch3_enum =
+ SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
+ 3, sta32x_limiter_select);
+static const struct soc_enum sta32x_limiter1_attack_rate_enum =
+ SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT,
+ 16, sta32x_limiter_attack_rate);
+static const struct soc_enum sta32x_limiter2_attack_rate_enum =
+ SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT,
+ 16, sta32x_limiter_attack_rate);
+static const struct soc_enum sta32x_limiter1_release_rate_enum =
+ SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT,
+ 16, sta32x_limiter_release_rate);
+static const struct soc_enum sta32x_limiter2_release_rate_enum =
+ SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT,
+ 16, sta32x_limiter_release_rate);
+static const struct snd_kcontrol_new sta32x_snd_controls[] = {
+SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv),
+SOC_SINGLE("Master Switch", STA32X_MMUTE, 0, 1, 1),
+SOC_SINGLE("Ch1 Switch", STA32X_MMUTE, 1, 1, 1),
+SOC_SINGLE("Ch2 Switch", STA32X_MMUTE, 2, 1, 1),
+SOC_SINGLE("Ch3 Switch", STA32X_MMUTE, 3, 1, 1),
+SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv),
+SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv),
+SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv),
+SOC_SINGLE("De-emphasis Filter Switch", STA32X_CONFD, STA32X_CONFD_DEMP_SHIFT, 1, 0),
+SOC_ENUM("Compressor/Limiter Switch", sta32x_drc_ac_enum),
+SOC_SINGLE("Miami Mode Switch", STA32X_CONFD, STA32X_CONFD_MME_SHIFT, 1, 0),
+SOC_SINGLE("Zero Cross Switch", STA32X_CONFE, STA32X_CONFE_ZCE_SHIFT, 1, 0),
+SOC_SINGLE("Soft Ramp Switch", STA32X_CONFE, STA32X_CONFE_SVE_SHIFT, 1, 0),
+SOC_SINGLE("Auto-Mute Switch", STA32X_CONFF, STA32X_CONFF_IDE_SHIFT, 1, 0),
+SOC_ENUM("Automode EQ", sta32x_auto_eq_enum),
+SOC_ENUM("Automode GC", sta32x_auto_gc_enum),
+SOC_ENUM("Automode XO", sta32x_auto_xo_enum),
+SOC_ENUM("Preset EQ", sta32x_preset_eq_enum),
+SOC_SINGLE("Ch1 Tone Control Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0),
+SOC_SINGLE("Ch2 Tone Control Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0),
+SOC_SINGLE("Ch1 EQ Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch2 EQ Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch1 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch2 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch3 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0),
+SOC_ENUM("Ch1 Limiter Select", sta32x_limiter_ch1_enum),
+SOC_ENUM("Ch2 Limiter Select", sta32x_limiter_ch2_enum),
+SOC_ENUM("Ch3 Limiter Select", sta32x_limiter_ch3_enum),
+SOC_SINGLE_TLV("Bass Tone Control", STA32X_TONE, STA32X_TONE_BTC_SHIFT, 15, 0, tone_tlv),
+SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, tone_tlv),
+SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum),
+SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum),
+SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
+SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
+
+/* depending on mode, the attack/release thresholds have
+ * two different enum definitions; provide both
+ */
+SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT,
+ 16, 0, sta32x_limiter_ac_attack_tlv),
+SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT,
+ 16, 0, sta32x_limiter_ac_attack_tlv),
+SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT,
+ 16, 0, sta32x_limiter_ac_release_tlv),
+SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT,
+ 16, 0, sta32x_limiter_ac_release_tlv),
+SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT,
+ 16, 0, sta32x_limiter_drc_attack_tlv),
+SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT,
+ 16, 0, sta32x_limiter_drc_attack_tlv),
+SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT,
+ 16, 0, sta32x_limiter_drc_release_tlv),
+SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT,
+ 16, 0, sta32x_limiter_drc_release_tlv),
+};
+
+static const struct snd_soc_dapm_widget sta32x_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("LEFT"),
+SND_SOC_DAPM_OUTPUT("RIGHT"),
+SND_SOC_DAPM_OUTPUT("SUB"),
+};
+
+static const struct snd_soc_dapm_route sta32x_dapm_routes[] = {
+ { "LEFT", NULL, "DAC" },
+ { "RIGHT", NULL, "DAC" },
+ { "SUB", NULL, "DAC" },
+};
+
+/* MCLK interpolation ratio per fs */
+static struct {
+ int fs;
+ int ir;
+} interpolation_ratios[] = {
+ { 32000, 0 },
+ { 44100, 0 },
+ { 48000, 0 },
+ { 88200, 1 },
+ { 96000, 1 },
+ { 176400, 2 },
+ { 192000, 2 },
+};
+
+/* MCLK to fs clock ratios */
+static struct {
+ int ratio;
+ int mcs;
+} mclk_ratios[3][7] = {
+ { { 768, 0 }, { 512, 1 }, { 384, 2 }, { 256, 3 },
+ { 128, 4 }, { 576, 5 }, { 0, 0 } },
+ { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } },
+ { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } },
+};
+
+
+/**
+ * sta32x_set_dai_sysclk - configure MCLK
+ * @codec_dai: the codec DAI
+ * @clk_id: the clock ID (ignored)
+ * @freq: the MCLK input frequency
+ * @dir: the clock direction (ignored)
+ *
+ * The value of MCLK is used to determine which sample rates are supported
+ * by the STA32X, based on the mclk_ratios table.
+ *
+ * This function must be called by the machine driver's 'startup' function,
+ * otherwise the list of supported sample rates will not be available in
+ * time for ALSA.
+ *
+ * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause
+ * theoretically possible sample rates to be enabled. Call it again with a
+ * proper value set one the external clock is set (most probably you would do
+ * that from a machine's driver 'hw_param' hook.
+ */
+static int sta32x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ int i, j, ir, fs;
+ unsigned int rates = 0;
+ unsigned int rate_min = -1;
+ unsigned int rate_max = 0;
+
+ pr_debug("mclk=%u\n", freq);
+ sta32x->mclk = freq;
+
+ if (sta32x->mclk) {
+ for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) {
+ ir = interpolation_ratios[i].ir;
+ fs = interpolation_ratios[i].fs;
+ for (j = 0; mclk_ratios[ir][j].ratio; j++) {
+ if (mclk_ratios[ir][j].ratio * fs == freq) {
+ rates |= snd_pcm_rate_to_rate_bit(fs);
+ if (fs < rate_min)
+ rate_min = fs;
+ if (fs > rate_max)
+ rate_max = fs;
+ }
+ }
+ }
+ /* FIXME: soc should support a rate list */
+ rates &= ~SNDRV_PCM_RATE_KNOT;
+
+ if (!rates) {
+ dev_err(codec->dev, "could not find a valid sample rate\n");
+ return -EINVAL;
+ }
+ } else {
+ /* enable all possible rates */
+ rates = STA32X_RATES;
+ rate_min = 32000;
+ rate_max = 192000;
+ }
+
+ codec_dai->driver->playback.rates = rates;
+ codec_dai->driver->playback.rate_min = rate_min;
+ codec_dai->driver->playback.rate_max = rate_max;
+ return 0;
+}
+
+/**
+ * sta32x_set_dai_fmt - configure the codec for the selected audio format
+ * @codec_dai: the codec DAI
+ * @fmt: a SND_SOC_DAIFMT_x value indicating the data format
+ *
+ * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the
+ * codec accordingly.
+ */
+static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ u8 confb = snd_soc_read(codec, STA32X_CONFB);
+
+ pr_debug("\n");
+ confb &= ~(STA32X_CONFB_C1IM | STA32X_CONFB_C2IM);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ sta32x->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ confb |= STA32X_CONFB_C2IM;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ confb |= STA32X_CONFB_C1IM;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, STA32X_CONFB, confb);
+ return 0;
+}
+
+/**
+ * sta32x_hw_params - program the STA32X with the given hardware parameters.
+ * @substream: the audio stream
+ * @params: the hardware parameters to set
+ * @dai: the SOC DAI (ignored)
+ *
+ * This function programs the hardware with the values provided.
+ * Specifically, the sample rate and the data format.
+ */
+static int sta32x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ unsigned int rate;
+ int i, mcs = -1, ir = -1;
+ u8 confa, confb;
+
+ rate = params_rate(params);
+ pr_debug("rate: %u\n", rate);
+ for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++)
+ if (interpolation_ratios[i].fs == rate)
+ ir = interpolation_ratios[i].ir;
+ if (ir < 0)
+ return -EINVAL;
+ for (i = 0; mclk_ratios[ir][i].ratio; i++)
+ if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk)
+ mcs = mclk_ratios[ir][i].mcs;
+ if (mcs < 0)
+ return -EINVAL;
+
+ confa = snd_soc_read(codec, STA32X_CONFA);
+ confa &= ~(STA32X_CONFA_MCS_MASK | STA32X_CONFA_IR_MASK);
+ confa |= (ir << STA32X_CONFA_IR_SHIFT) | (mcs << STA32X_CONFA_MCS_SHIFT);
+
+ confb = snd_soc_read(codec, STA32X_CONFB);
+ confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB);
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S24_LE:
+ case SNDRV_PCM_FORMAT_S24_BE:
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ case SNDRV_PCM_FORMAT_S24_3BE:
+ pr_debug("24bit\n");
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32_LE:
+ case SNDRV_PCM_FORMAT_S32_BE:
+ pr_debug("24bit or 32bit\n");
+ switch (sta32x->format) {
+ case SND_SOC_DAIFMT_I2S:
+ confb |= 0x0;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ confb |= 0x1;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ confb |= 0x2;
+ break;
+ }
+
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ case SNDRV_PCM_FORMAT_S20_3BE:
+ pr_debug("20bit\n");
+ switch (sta32x->format) {
+ case SND_SOC_DAIFMT_I2S:
+ confb |= 0x4;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ confb |= 0x5;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ confb |= 0x6;
+ break;
+ }
+
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ case SNDRV_PCM_FORMAT_S18_3BE:
+ pr_debug("18bit\n");
+ switch (sta32x->format) {
+ case SND_SOC_DAIFMT_I2S:
+ confb |= 0x8;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ confb |= 0x9;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ confb |= 0xa;
+ break;
+ }
+
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ case SNDRV_PCM_FORMAT_S16_BE:
+ pr_debug("16bit\n");
+ switch (sta32x->format) {
+ case SND_SOC_DAIFMT_I2S:
+ confb |= 0x0;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ confb |= 0xd;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ confb |= 0xe;
+ break;
+ }
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, STA32X_CONFA, confa);
+ snd_soc_write(codec, STA32X_CONFB, confb);
+ return 0;
+}
+
+/**
+ * sta32x_set_bias_level - DAPM callback
+ * @codec: the codec device
+ * @level: DAPM power level
+ *
+ * This is called by ALSA to put the codec into low power mode
+ * or to wake it up. If the codec is powered off completely
+ * all registers must be restored after power on.
+ */
+static int sta32x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ int ret;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("level = %d\n", level);
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ /* Full power on */
+ snd_soc_update_bits(codec, STA32X_CONFF,
+ STA32X_CONFF_PWDN | STA32X_CONFF_EAPD,
+ STA32X_CONFF_PWDN | STA32X_CONFF_EAPD);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
+ sta32x->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ snd_soc_cache_sync(codec);
+ }
+
+ /* Power up to mute */
+ /* FIXME */
+ snd_soc_update_bits(codec, STA32X_CONFF,
+ STA32X_CONFF_PWDN | STA32X_CONFF_EAPD,
+ STA32X_CONFF_PWDN | STA32X_CONFF_EAPD);
+
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ /* The chip runs through the power down sequence for us. */
+ snd_soc_update_bits(codec, STA32X_CONFF,
+ STA32X_CONFF_PWDN | STA32X_CONFF_EAPD,
+ STA32X_CONFF_PWDN);
+ msleep(300);
+
+ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies),
+ sta32x->supplies);
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+static struct snd_soc_dai_ops sta32x_dai_ops = {
+ .hw_params = sta32x_hw_params,
+ .set_sysclk = sta32x_set_dai_sysclk,
+ .set_fmt = sta32x_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver sta32x_dai = {
+ .name = "STA32X",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = STA32X_RATES,
+ .formats = STA32X_FORMATS,
+ },
+ .ops = &sta32x_dai_ops,
+};
+
+#ifdef CONFIG_PM
+static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int sta32x_resume(struct snd_soc_codec *codec)
+{
+ sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+#else
+#define sta32x_suspend NULL
+#define sta32x_resume NULL
+#endif
+
+static int sta32x_probe(struct snd_soc_codec *codec)
+{
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ int i, ret = 0;
+
+ sta32x->codec = codec;
+
+ /* regulators */
+ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++)
+ sta32x->supplies[i].supply = sta32x_supply_names[i];
+
+ ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies),
+ sta32x->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+ goto err;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
+ sta32x->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+ goto err_get;
+ }
+
+ /* Tell ASoC what kind of I/O to use to read the registers. ASoC will
+ * then do the I2C transactions itself.
+ */
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret);
+ return ret;
+ }
+
+ /* read reg reset values into cache */
+ for (i = 0; i < STA32X_REGISTER_COUNT; i++)
+ snd_soc_cache_write(codec, i, sta32x_regs[i]);
+
+ /* FIXME enable thermal warning adjustment and recovery */
+ snd_soc_update_bits(codec, STA32X_CONFA,
+ STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0);
+
+ /* FIXME select 2.1 mode */
+ snd_soc_update_bits(codec, STA32X_CONFF,
+ STA32X_CONFF_OCFG_MASK,
+ 1 << STA32X_CONFF_OCFG_SHIFT);
+
+ /* FIXME channel to output mapping */
+ snd_soc_update_bits(codec, STA32X_C1CFG,
+ STA32X_CxCFG_OM_MASK,
+ 0 << STA32X_CxCFG_OM_SHIFT);
+ snd_soc_update_bits(codec, STA32X_C2CFG,
+ STA32X_CxCFG_OM_MASK,
+ 1 << STA32X_CxCFG_OM_SHIFT);
+ snd_soc_update_bits(codec, STA32X_C3CFG,
+ STA32X_CxCFG_OM_MASK,
+ 2 << STA32X_CxCFG_OM_SHIFT);
+
+ sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ /* Bias level configuration will have done an extra enable */
+ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+ return 0;
+
+err_get:
+ regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+err:
+ return ret;
+}
+
+static int sta32x_remove(struct snd_soc_codec *codec)
+{
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+
+ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+ regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+ return 0;
+}
+
+static const struct snd_soc_codec_driver sta32x_codec = {
+ .probe = sta32x_probe,
+ .remove = sta32x_remove,
+ .suspend = sta32x_suspend,
+ .resume = sta32x_resume,
+ .reg_cache_size = STA32X_REGISTER_COUNT,
+ .reg_word_size = sizeof(u8),
+ .set_bias_level = sta32x_set_bias_level,
+ .controls = sta32x_snd_controls,
+ .num_controls = ARRAY_SIZE(sta32x_snd_controls),
+ .dapm_widgets = sta32x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sta32x_dapm_widgets),
+ .dapm_routes = sta32x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(sta32x_dapm_routes),
+};
+
+static __devinit int sta32x_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct sta32x_priv *sta32x;
+ int ret;
+
+ sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL);
+ if (!sta32x)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, sta32x);
+
+ ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static __devexit int sta32x_i2c_remove(struct i2c_client *client)
+{
+ struct sta32x_priv *sta32x = i2c_get_clientdata(client);
+ struct snd_soc_codec *codec = sta32x->codec;
+
+ if (codec)
+ sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+ if (codec) {
+ snd_soc_unregister_codec(&client->dev);
+ snd_soc_codec_set_drvdata(codec, NULL);
+ }
+
+ kfree(sta32x);
+ return 0;
+}
+
+static const struct i2c_device_id sta32x_i2c_id[] = {
+ { "sta326", 0 },
+ { "sta328", 0 },
+ { "sta329", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id);
+
+static struct i2c_driver sta32x_i2c_driver = {
+ .driver = {
+ .name = "sta32x",
+ .owner = THIS_MODULE,
+ },
+ .probe = sta32x_i2c_probe,
+ .remove = __devexit_p(sta32x_i2c_remove),
+ .id_table = sta32x_i2c_id,
+};
+
+static int __init sta32x_init(void)
+{
+ return i2c_add_driver(&sta32x_i2c_driver);
+}
+module_init(sta32x_init);
+
+static void __exit sta32x_exit(void)
+{
+ i2c_del_driver(&sta32x_i2c_driver);
+}
+module_exit(sta32x_exit);
+
+MODULE_DESCRIPTION("ASoC STA32X driver");
+MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h
new file mode 100644
index 0000000..b97ee5a
--- /dev/null
+++ b/sound/soc/codecs/sta32x.h
@@ -0,0 +1,210 @@
+/*
+ * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on code from:
+ * Wolfson Microelectronics PLC.
+ * Mark Brown <broonie@opensource.wolfsonmicro.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.
+ */
+#ifndef _ASOC_STA_32X_H
+#define _ASOC_STA_32X_H
+
+/* STA326 register addresses */
+
+#define STA32X_REGISTER_COUNT 0x2d
+
+#define STA32X_CONFA 0x00
+#define STA32X_CONFB 0x01
+#define STA32X_CONFC 0x02
+#define STA32X_CONFD 0x03
+#define STA32X_CONFE 0x04
+#define STA32X_CONFF 0x05
+#define STA32X_MMUTE 0x06
+#define STA32X_MVOL 0x07
+#define STA32X_C1VOL 0x08
+#define STA32X_C2VOL 0x09
+#define STA32X_C3VOL 0x0a
+#define STA32X_AUTO1 0x0b
+#define STA32X_AUTO2 0x0c
+#define STA32X_AUTO3 0x0d
+#define STA32X_C1CFG 0x0e
+#define STA32X_C2CFG 0x0f
+#define STA32X_C3CFG 0x10
+#define STA32X_TONE 0x11
+#define STA32X_L1AR 0x12
+#define STA32X_L1ATRT 0x13
+#define STA32X_L2AR 0x14
+#define STA32X_L2ATRT 0x15
+#define STA32X_CFADDR2 0x16
+#define STA32X_B1CF1 0x17
+#define STA32X_B1CF2 0x18
+#define STA32X_B1CF3 0x19
+#define STA32X_B2CF1 0x1a
+#define STA32X_B2CF2 0x1b
+#define STA32X_B2CF3 0x1c
+#define STA32X_A1CF1 0x1d
+#define STA32X_A1CF2 0x1e
+#define STA32X_A1CF3 0x1f
+#define STA32X_A2CF1 0x20
+#define STA32X_A2CF2 0x21
+#define STA32X_A2CF3 0x22
+#define STA32X_B0CF1 0x23
+#define STA32X_B0CF2 0x24
+#define STA32X_B0CF3 0x25
+#define STA32X_CFUD 0x26
+#define STA32X_MPCC1 0x27
+#define STA32X_MPCC2 0x28
+/* Reserved 0x29 */
+/* Reserved 0x2a */
+#define STA32X_Reserved 0x2a
+#define STA32X_FDRC1 0x2b
+#define STA32X_FDRC2 0x2c
+/* Reserved 0x2d */
+
+
+/* STA326 register field definitions */
+
+/* 0x00 CONFA */
+#define STA32X_CONFA_MCS_MASK 0x03
+#define STA32X_CONFA_MCS_SHIFT 0
+#define STA32X_CONFA_IR_MASK 0x18
+#define STA32X_CONFA_IR_SHIFT 3
+#define STA32X_CONFA_TWRB 0x20
+#define STA32X_CONFA_TWAB 0x40
+#define STA32X_CONFA_FDRB 0x80
+
+/* 0x01 CONFB */
+#define STA32X_CONFB_SAI_MASK 0x0f
+#define STA32X_CONFB_SAI_SHIFT 0
+#define STA32X_CONFB_SAIFB 0x10
+#define STA32X_CONFB_DSCKE 0x20
+#define STA32X_CONFB_C1IM 0x40
+#define STA32X_CONFB_C2IM 0x80
+
+/* 0x02 CONFC */
+#define STA32X_CONFC_OM_MASK 0x03
+#define STA32X_CONFC_OM_SHIFT 0
+#define STA32X_CONFC_CSZ_MASK 0x7c
+#define STA32X_CONFC_CSZ_SHIFT 2
+
+/* 0x03 CONFD */
+#define STA32X_CONFD_HPB 0x01
+#define STA32X_CONFD_HPB_SHIFT 0
+#define STA32X_CONFD_DEMP 0x02
+#define STA32X_CONFD_DEMP_SHIFT 1
+#define STA32X_CONFD_DSPB 0x04
+#define STA32X_CONFD_DSPB_SHIFT 2
+#define STA32X_CONFD_PSL 0x08
+#define STA32X_CONFD_PSL_SHIFT 3
+#define STA32X_CONFD_BQL 0x10
+#define STA32X_CONFD_BQL_SHIFT 4
+#define STA32X_CONFD_DRC 0x20
+#define STA32X_CONFD_DRC_SHIFT 5
+#define STA32X_CONFD_ZDE 0x40
+#define STA32X_CONFD_ZDE_SHIFT 6
+#define STA32X_CONFD_MME 0x80
+#define STA32X_CONFD_MME_SHIFT 7
+
+/* 0x04 CONFE */
+#define STA32X_CONFE_MPCV 0x01
+#define STA32X_CONFE_MPCV_SHIFT 0
+#define STA32X_CONFE_MPC 0x02
+#define STA32X_CONFE_MPC_SHIFT 1
+#define STA32X_CONFE_AME 0x08
+#define STA32X_CONFE_AME_SHIFT 3
+#define STA32X_CONFE_PWMS 0x10
+#define STA32X_CONFE_PWMS_SHIFT 4
+#define STA32X_CONFE_ZCE 0x40
+#define STA32X_CONFE_ZCE_SHIFT 6
+#define STA32X_CONFE_SVE 0x80
+#define STA32X_CONFE_SVE_SHIFT 7
+
+/* 0x05 CONFF */
+#define STA32X_CONFF_OCFG_MASK 0x03
+#define STA32X_CONFF_OCFG_SHIFT 0
+#define STA32X_CONFF_IDE 0x04
+#define STA32X_CONFF_IDE_SHIFT 3
+#define STA32X_CONFF_BCLE 0x08
+#define STA32X_CONFF_ECLE 0x20
+#define STA32X_CONFF_PWDN 0x40
+#define STA32X_CONFF_EAPD 0x80
+
+/* 0x06 MMUTE */
+#define STA32X_MMUTE_MMUTE 0x01
+
+/* 0x0b AUTO1 */
+#define STA32X_AUTO1_AMEQ_MASK 0x03
+#define STA32X_AUTO1_AMEQ_SHIFT 0
+#define STA32X_AUTO1_AMV_MASK 0xc0
+#define STA32X_AUTO1_AMV_SHIFT 2
+#define STA32X_AUTO1_AMGC_MASK 0x30
+#define STA32X_AUTO1_AMGC_SHIFT 4
+#define STA32X_AUTO1_AMPS 0x80
+
+/* 0x0c AUTO2 */
+#define STA32X_AUTO2_AMAME 0x01
+#define STA32X_AUTO2_AMAM_MASK 0x0e
+#define STA32X_AUTO2_AMAM_SHIFT 1
+#define STA32X_AUTO2_XO_MASK 0xf0
+#define STA32X_AUTO2_XO_SHIFT 4
+
+/* 0x0d AUTO3 */
+#define STA32X_AUTO3_PEQ_MASK 0x1f
+#define STA32X_AUTO3_PEQ_SHIFT 0
+
+/* 0x0e 0x0f 0x10 CxCFG */
+#define STA32X_CxCFG_TCB 0x01 /* only C1 and C2 */
+#define STA32X_CxCFG_TCB_SHIFT 0
+#define STA32X_CxCFG_EQBP 0x02 /* only C1 and C2 */
+#define STA32X_CxCFG_EQBP_SHIFT 1
+#define STA32X_CxCFG_VBP 0x03
+#define STA32X_CxCFG_VBP_SHIFT 2
+#define STA32X_CxCFG_BO 0x04
+#define STA32X_CxCFG_LS_MASK 0x30
+#define STA32X_CxCFG_LS_SHIFT 4
+#define STA32X_CxCFG_OM_MASK 0xc0
+#define STA32X_CxCFG_OM_SHIFT 6
+
+/* 0x11 TONE */
+#define STA32X_TONE_BTC_SHIFT 0
+#define STA32X_TONE_TTC_SHIFT 4
+
+/* 0x12 0x13 0x14 0x15 limiter attack/release */
+#define STA32X_LxA_SHIFT 0
+#define STA32X_LxR_SHIFT 4
+
+/* 0x26 CFUD */
+#define STA32X_CFUD_W1 0x01
+#define STA32X_CFUD_WA 0x02
+#define STA32X_CFUD_R1 0x04
+#define STA32X_CFUD_RA 0x08
+
+
+/* biquad filter coefficient table offsets */
+#define STA32X_C1_BQ_BASE 0
+#define STA32X_C2_BQ_BASE 20
+#define STA32X_CH_BQ_NUM 4
+#define STA32X_BQ_NUM_COEF 5
+#define STA32X_XO_HP_BQ_BASE 40
+#define STA32X_XO_LP_BQ_BASE 45
+#define STA32X_C1_PRESCALE 50
+#define STA32X_C2_PRESCALE 51
+#define STA32X_C1_POSTSCALE 52
+#define STA32X_C2_POSTSCALE 53
+#define STA32X_C3_POSTSCALE 54
+#define STA32X_TW_POSTSCALE 55
+#define STA32X_C1_MIX1 56
+#define STA32X_C1_MIX2 57
+#define STA32X_C2_MIX1 58
+#define STA32X_C2_MIX2 59
+#define STA32X_C3_MIX1 60
+#define STA32X_C3_MIX2 61
+
+#endif /* _ASOC_STA_32X_H */
diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c
new file mode 100644
index 0000000..a2a09f8
--- /dev/null
+++ b/sound/soc/codecs/wm8782.c
@@ -0,0 +1,80 @@
+/*
+ * sound/soc/codecs/wm8782.c
+ * simple, strap-pin configured 24bit 2ch ADC
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on ad73311.c
+ * Copyright: Analog Device Inc.
+ * Author: Cliff Cai <cliff.cai@analog.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/init.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+static struct snd_soc_dai_driver wm8782_dai = {
+ .name = "wm8782",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ /* For configurations with FSAMPEN=0 */
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ },
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8782;
+
+static __devinit int wm8782_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev,
+ &soc_codec_dev_wm8782, &wm8782_dai, 1);
+}
+
+static int __devexit wm8782_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver wm8782_codec_driver = {
+ .driver = {
+ .name = "wm8782",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8782_probe,
+ .remove = wm8782_remove,
+};
+
+static int __init wm8782_init(void)
+{
+ return platform_driver_register(&wm8782_codec_driver);
+}
+module_init(wm8782_init);
+
+static void __exit wm8782_exit(void)
+{
+ platform_driver_unregister(&wm8782_codec_driver);
+}
+module_exit(wm8782_exit);
+
+MODULE_DESCRIPTION("ASoC WM8782 driver");
+MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 9b3bba4..b085575 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2560,6 +2560,7 @@
static const struct i2c_device_id wm8904_i2c_id[] = {
{ "wm8904", WM8904 },
{ "wm8912", WM8912 },
+ { "wm8918", WM8904 }, /* Actually a subset, updates to follow */
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id);
diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c
index e2ab4fa..423baa9 100644
--- a/sound/soc/codecs/wm8915.c
+++ b/sound/soc/codecs/wm8915.c
@@ -41,14 +41,12 @@
#define HPOUT2L 4
#define HPOUT2R 8
-#define WM8915_NUM_SUPPLIES 6
+#define WM8915_NUM_SUPPLIES 4
static const char *wm8915_supply_names[WM8915_NUM_SUPPLIES] = {
- "DCVDD",
"DBVDD",
"AVDD1",
"AVDD2",
"CPVDD",
- "MICVDD",
};
struct wm8915_priv {
@@ -57,6 +55,7 @@
int ldo1ena;
int sysclk;
+ int sysclk_src;
int fll_src;
int fll_fref;
@@ -76,6 +75,7 @@
struct wm8915_pdata pdata;
int rx_rate[WM8915_AIFS];
+ int bclk_rate[WM8915_AIFS];
/* Platform dependant ReTune mobile configuration */
int num_retune_mobile_texts;
@@ -113,8 +113,6 @@
WM8915_REGULATOR_EVENT(1)
WM8915_REGULATOR_EVENT(2)
WM8915_REGULATOR_EVENT(3)
-WM8915_REGULATOR_EVENT(4)
-WM8915_REGULATOR_EVENT(5)
static const u16 wm8915_reg[WM8915_MAX_REGISTER] = {
[WM8915_SOFTWARE_RESET] = 0x8915,
@@ -1565,6 +1563,50 @@
return snd_soc_write(codec, WM8915_SOFTWARE_RESET, 0x8915);
}
+static const int bclk_divs[] = {
+ 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
+};
+
+static void wm8915_update_bclk(struct snd_soc_codec *codec)
+{
+ struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
+ int aif, best, cur_val, bclk_rate, bclk_reg, i;
+
+ /* Don't bother if we're in a low frequency idle mode that
+ * can't support audio.
+ */
+ if (wm8915->sysclk < 64000)
+ return;
+
+ for (aif = 0; aif < WM8915_AIFS; aif++) {
+ switch (aif) {
+ case 0:
+ bclk_reg = WM8915_AIF1_BCLK;
+ break;
+ case 1:
+ bclk_reg = WM8915_AIF2_BCLK;
+ break;
+ }
+
+ bclk_rate = wm8915->bclk_rate[aif];
+
+ /* Pick a divisor for BCLK as close as we can get to ideal */
+ best = 0;
+ for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
+ cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate;
+ if (cur_val < 0) /* BCLK table is sorted */
+ break;
+ best = i;
+ }
+ bclk_rate = wm8915->sysclk / bclk_divs[best];
+ dev_dbg(codec->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n",
+ bclk_divs[best], bclk_rate);
+
+ snd_soc_update_bits(codec, bclk_reg,
+ WM8915_AIF1_BCLK_DIV_MASK, best);
+ }
+}
+
static int wm8915_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
@@ -1717,10 +1759,6 @@
return 0;
}
-static const int bclk_divs[] = {
- 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
-};
-
static const int dsp_divs[] = {
48000, 32000, 16000, 8000
};
@@ -1731,17 +1769,11 @@
{
struct snd_soc_codec *codec = dai->codec;
struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
- int bits, i, bclk_rate, best, cur_val;
+ int bits, i, bclk_rate;
int aifdata = 0;
- int bclk = 0;
int lrclk = 0;
int dsp = 0;
- int aifdata_reg, bclk_reg, lrclk_reg, dsp_shift;
-
- if (!wm8915->sysclk) {
- dev_err(codec->dev, "SYSCLK not configured\n");
- return -EINVAL;
- }
+ int aifdata_reg, lrclk_reg, dsp_shift;
switch (dai->id) {
case 0:
@@ -1753,7 +1785,6 @@
aifdata_reg = WM8915_AIF1TX_DATA_CONFIGURATION_1;
lrclk_reg = WM8915_AIF1_TX_LRCLK_1;
}
- bclk_reg = WM8915_AIF1_BCLK;
dsp_shift = 0;
break;
case 1:
@@ -1765,7 +1796,6 @@
aifdata_reg = WM8915_AIF2TX_DATA_CONFIGURATION_1;
lrclk_reg = WM8915_AIF2_TX_LRCLK_1;
}
- bclk_reg = WM8915_AIF2_BCLK;
dsp_shift = WM8915_DSP2_DIV_SHIFT;
break;
default:
@@ -1779,6 +1809,9 @@
return bclk_rate;
}
+ wm8915->bclk_rate[dai->id] = bclk_rate;
+ wm8915->rx_rate[dai->id] = params_rate(params);
+
/* Needs looking at for TDM */
bits = snd_pcm_format_width(params_format(params));
if (bits < 0)
@@ -1796,18 +1829,7 @@
}
dsp |= i << dsp_shift;
- /* Pick a divisor for BCLK as close as we can get to ideal */
- best = 0;
- for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
- cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate;
- if (cur_val < 0) /* BCLK table is sorted */
- break;
- best = i;
- }
- bclk_rate = wm8915->sysclk / bclk_divs[best];
- dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n",
- bclk_divs[best], bclk_rate);
- bclk |= best;
+ wm8915_update_bclk(codec);
lrclk = bclk_rate / params_rate(params);
dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
@@ -1817,14 +1839,11 @@
WM8915_AIF1TX_WL_MASK |
WM8915_AIF1TX_SLOT_LEN_MASK,
aifdata);
- snd_soc_update_bits(codec, bclk_reg, WM8915_AIF1_BCLK_DIV_MASK, bclk);
snd_soc_update_bits(codec, lrclk_reg, WM8915_AIF1RX_RATE_MASK,
lrclk);
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_2,
WM8915_DSP1_DIV_SHIFT << dsp_shift, dsp);
- wm8915->rx_rate[dai->id] = params_rate(params);
-
return 0;
}
@@ -1838,6 +1857,9 @@
int src;
int old;
+ if (freq == wm8915->sysclk && clk_id == wm8915->sysclk_src)
+ return 0;
+
/* Disable SYSCLK while we reconfigure */
old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA;
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
@@ -1882,6 +1904,8 @@
return -EINVAL;
}
+ wm8915_update_bclk(codec);
+
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
WM8915_SYSCLK_SRC_MASK | WM8915_SYSCLK_DIV_MASK,
src << WM8915_SYSCLK_SRC_SHIFT | ratediv);
@@ -1889,6 +1913,8 @@
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
WM8915_SYSCLK_ENA, old);
+ wm8915->sysclk_src = clk_id;
+
return 0;
}
@@ -2007,6 +2033,7 @@
unsigned int Fref, unsigned int Fout)
{
struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
+ struct i2c_client *i2c = to_i2c_client(codec->dev);
struct _fll_div fll_div;
unsigned long timeout;
int ret, reg;
@@ -2093,7 +2120,18 @@
else
timeout = msecs_to_jiffies(2);
- wait_for_completion_timeout(&wm8915->fll_lock, timeout);
+ /* Allow substantially longer if we've actually got the IRQ */
+ if (i2c->irq)
+ timeout *= 1000;
+
+ ret = wait_for_completion_timeout(&wm8915->fll_lock, timeout);
+
+ if (ret == 0 && i2c->irq) {
+ dev_err(codec->dev, "Timed out waiting for FLL\n");
+ ret = -ETIMEDOUT;
+ } else {
+ ret = 0;
+ }
dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
@@ -2101,7 +2139,7 @@
wm8915->fll_fout = Fout;
wm8915->fll_src = source;
- return 0;
+ return ret;
}
#ifdef CONFIG_GPIOLIB
@@ -2293,6 +2331,12 @@
SND_JACK_HEADSET | SND_JACK_BTN_0);
wm8915->jack_mic = true;
wm8915->detecting = false;
+
+ /* Increase poll rate to give better responsiveness
+ * for buttons */
+ snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
+ WM8915_MICD_RATE_MASK,
+ 5 << WM8915_MICD_RATE_SHIFT);
}
/* If we detected a lower impedence during initial startup
@@ -2333,15 +2377,17 @@
SND_JACK_HEADPHONE,
SND_JACK_HEADSET |
SND_JACK_BTN_0);
+
+ /* Increase the detection rate a bit for
+ * responsiveness.
+ */
+ snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
+ WM8915_MICD_RATE_MASK,
+ 7 << WM8915_MICD_RATE_SHIFT);
+
wm8915->detecting = false;
}
}
-
- /* Increase poll rate to give better responsiveness for buttons */
- if (!wm8915->detecting)
- snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
- WM8915_MICD_RATE_MASK,
- 5 << WM8915_MICD_RATE_SHIFT);
}
static irqreturn_t wm8915_irq(int irq, void *data)
@@ -2383,6 +2429,20 @@
}
}
+static irqreturn_t wm8915_edge_irq(int irq, void *data)
+{
+ irqreturn_t ret = IRQ_NONE;
+ irqreturn_t val;
+
+ do {
+ val = wm8915_irq(irq, data);
+ if (val != IRQ_NONE)
+ ret = val;
+ } while (val != IRQ_NONE);
+
+ return ret;
+}
+
static void wm8915_retune_mobile_pdata(struct snd_soc_codec *codec)
{
struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
@@ -2482,8 +2542,6 @@
wm8915->disable_nb[1].notifier_call = wm8915_regulator_event_1;
wm8915->disable_nb[2].notifier_call = wm8915_regulator_event_2;
wm8915->disable_nb[3].notifier_call = wm8915_regulator_event_3;
- wm8915->disable_nb[4].notifier_call = wm8915_regulator_event_4;
- wm8915->disable_nb[5].notifier_call = wm8915_regulator_event_5;
/* This should really be moved into the regulator core */
for (i = 0; i < ARRAY_SIZE(wm8915->supplies); i++) {
@@ -2709,8 +2767,14 @@
irq_flags |= IRQF_ONESHOT;
- ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq,
- irq_flags, "wm8915", codec);
+ if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING))
+ ret = request_threaded_irq(i2c->irq, NULL,
+ wm8915_edge_irq,
+ irq_flags, "wm8915", codec);
+ else
+ ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq,
+ irq_flags, "wm8915", codec);
+
if (ret == 0) {
/* Unmask the interrupt */
snd_soc_update_bits(codec, WM8915_INTERRUPT_CONTROL,
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 25580e3..056daa0 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -297,8 +297,6 @@
if (ret)
goto error_ret;
ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- if (ret)
- goto error_ret;
error_ret:
return ret;
@@ -683,8 +681,6 @@
}
}
ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- if (ret)
- goto error_ret;
error_ret:
return ret;
@@ -730,9 +726,6 @@
if (ret)
return ret;
ret = wm8940_add_widgets(codec);
- if (ret)
- return ret;
-
return ret;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 5e05eed..8499c56 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -78,6 +78,8 @@
#ifdef CONFIG_GPIOLIB
struct gpio_chip gpio_chip;
#endif
+
+ int irq;
};
/* We can't use the same notifier block for more than one supply and
@@ -1982,6 +1984,7 @@
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
/* The VU bits for the headphones are in a different register to the mute
* bits and only take effect on the PGA if it is actually powered.
@@ -2119,6 +2122,18 @@
SOC_SINGLE_TLV("Speaker Boost Volume", WM8962_CLASS_D_CONTROL_2, 0, 7, 0,
classd_tlv),
+
+SOC_SINGLE("EQ Switch", WM8962_EQ1, WM8962_EQ_ENA_SHIFT, 1, 0),
+SOC_DOUBLE_R_TLV("EQ1 Volume", WM8962_EQ2, WM8962_EQ22,
+ WM8962_EQL_B1_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ2 Volume", WM8962_EQ2, WM8962_EQ22,
+ WM8962_EQL_B2_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ3 Volume", WM8962_EQ2, WM8962_EQ22,
+ WM8962_EQL_B3_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23,
+ WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23,
+ WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv),
};
static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = {
@@ -2184,6 +2199,8 @@
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
+ struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+ unsigned long timeout;
int src;
int fll;
@@ -2203,9 +2220,19 @@
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
- if (fll)
+ if (fll) {
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_ENA, WM8962_FLL_ENA);
+ if (wm8962->irq) {
+ timeout = msecs_to_jiffies(5);
+ timeout = wait_for_completion_timeout(&wm8962->fll_lock,
+ timeout);
+
+ if (timeout == 0)
+ dev_err(codec->dev,
+ "Timed out starting FLL\n");
+ }
+ }
break;
case SND_SOC_DAPM_POST_PMD:
@@ -2763,18 +2790,44 @@
1, -1, 2, 3, 4, -1, 6, 8, -1, 12, 16, 24, -1, 32, 32, 32
};
+static const int sysclk_rates[] = {
+ 64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536,
+};
+
static void wm8962_configure_bclk(struct snd_soc_codec *codec)
{
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
int dspclk, i;
int clocking2 = 0;
+ int clocking4 = 0;
int aif2 = 0;
- if (!wm8962->bclk) {
- dev_dbg(codec->dev, "No BCLK rate configured\n");
+ if (!wm8962->sysclk_rate) {
+ dev_dbg(codec->dev, "No SYSCLK configured\n");
return;
}
+ if (!wm8962->bclk || !wm8962->lrclk) {
+ dev_dbg(codec->dev, "No audio clocks configured\n");
+ return;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(sysclk_rates); i++) {
+ if (sysclk_rates[i] == wm8962->sysclk_rate / wm8962->lrclk) {
+ clocking4 |= i << WM8962_SYSCLK_RATE_SHIFT;
+ break;
+ }
+ }
+
+ if (i == ARRAY_SIZE(sysclk_rates)) {
+ dev_err(codec->dev, "Unsupported sysclk ratio %d\n",
+ wm8962->sysclk_rate / wm8962->lrclk);
+ return;
+ }
+
+ snd_soc_update_bits(codec, WM8962_CLOCKING_4,
+ WM8962_SYSCLK_RATE_MASK, clocking4);
+
dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
if (dspclk < 0) {
dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk);
@@ -2844,6 +2897,8 @@
/* VMID 2*50k */
snd_soc_update_bits(codec, WM8962_PWR_MGMT_1,
WM8962_VMID_SEL_MASK, 0x80);
+
+ wm8962_configure_bclk(codec);
break;
case SND_SOC_BIAS_STANDBY:
@@ -2876,8 +2931,6 @@
snd_soc_update_bits(codec, WM8962_CLOCKING2,
WM8962_CLKREG_OVD,
WM8962_CLKREG_OVD);
-
- wm8962_configure_bclk(codec);
}
/* VMID 2*250k */
@@ -2918,10 +2971,6 @@
{ 96000, 6 },
};
-static const int sysclk_rates[] = {
- 64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536,
-};
-
static int wm8962_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -2929,41 +2978,27 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
- int rate = params_rate(params);
int i;
int aif0 = 0;
int adctl3 = 0;
- int clocking4 = 0;
wm8962->bclk = snd_soc_params_to_bclk(params);
wm8962->lrclk = params_rate(params);
for (i = 0; i < ARRAY_SIZE(sr_vals); i++) {
- if (sr_vals[i].rate == rate) {
+ if (sr_vals[i].rate == wm8962->lrclk) {
adctl3 |= sr_vals[i].reg;
break;
}
}
if (i == ARRAY_SIZE(sr_vals)) {
- dev_err(codec->dev, "Unsupported rate %dHz\n", rate);
+ dev_err(codec->dev, "Unsupported rate %dHz\n", wm8962->lrclk);
return -EINVAL;
}
- if (rate % 8000 == 0)
+ if (wm8962->lrclk % 8000 == 0)
adctl3 |= WM8962_SAMPLE_RATE_INT_MODE;
- for (i = 0; i < ARRAY_SIZE(sysclk_rates); i++) {
- if (sysclk_rates[i] == wm8962->sysclk_rate / rate) {
- clocking4 |= i << WM8962_SYSCLK_RATE_SHIFT;
- break;
- }
- }
- if (i == ARRAY_SIZE(sysclk_rates)) {
- dev_err(codec->dev, "Unsupported sysclk ratio %d\n",
- wm8962->sysclk_rate / rate);
- return -EINVAL;
- }
-
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
@@ -2985,8 +3020,6 @@
snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_3,
WM8962_SAMPLE_RATE_INT_MODE |
WM8962_SAMPLE_RATE_MASK, adctl3);
- snd_soc_update_bits(codec, WM8962_CLOCKING_4,
- WM8962_SYSCLK_RATE_MASK, clocking4);
wm8962_configure_bclk(codec);
@@ -3261,16 +3294,31 @@
dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
- /* This should be a massive overestimate */
- timeout = msecs_to_jiffies(1);
+ ret = 0;
- wait_for_completion_timeout(&wm8962->fll_lock, timeout);
+ if (fll1 & WM8962_FLL_ENA) {
+ /* This should be a massive overestimate but go even
+ * higher if we'll error out
+ */
+ if (wm8962->irq)
+ timeout = msecs_to_jiffies(5);
+ else
+ timeout = msecs_to_jiffies(1);
+
+ timeout = wait_for_completion_timeout(&wm8962->fll_lock,
+ timeout);
+
+ if (timeout == 0 && wm8962->irq) {
+ dev_err(codec->dev, "FLL lock timed out");
+ ret = -ETIMEDOUT;
+ }
+ }
wm8962->fll_fref = Fref;
wm8962->fll_fout = Fout;
wm8962->fll_src = source;
- return 0;
+ return ret;
}
static int wm8962_mute(struct snd_soc_dai *dai, int mute)
@@ -3731,8 +3779,6 @@
int ret;
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
struct wm8962_pdata *pdata = dev_get_platdata(codec->dev);
- struct i2c_client *i2c = container_of(codec->dev, struct i2c_client,
- dev);
u16 *reg_cache = codec->reg_cache;
int i, trigger, irq_pol;
bool dmicclk, dmicdat;
@@ -3871,6 +3917,9 @@
snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME,
WM8962_HPOUT_VU, WM8962_HPOUT_VU);
+ /* Stereo control for EQ */
+ snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0);
+
wm8962_add_widgets(codec);
/* Save boards having to disable DMIC when not in use */
@@ -3899,7 +3948,7 @@
wm8962_init_beep(codec);
wm8962_init_gpio(codec);
- if (i2c->irq) {
+ if (wm8962->irq) {
if (pdata && pdata->irq_active_low) {
trigger = IRQF_TRIGGER_LOW;
irq_pol = WM8962_IRQ_POL;
@@ -3911,12 +3960,13 @@
snd_soc_update_bits(codec, WM8962_INTERRUPT_CONTROL,
WM8962_IRQ_POL, irq_pol);
- ret = request_threaded_irq(i2c->irq, NULL, wm8962_irq,
+ ret = request_threaded_irq(wm8962->irq, NULL, wm8962_irq,
trigger | IRQF_ONESHOT,
"wm8962", codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to request IRQ %d: %d\n",
- i2c->irq, ret);
+ wm8962->irq, ret);
+ wm8962->irq = 0;
/* Non-fatal */
} else {
/* Enable some IRQs by default */
@@ -3941,12 +3991,10 @@
static int wm8962_remove(struct snd_soc_codec *codec)
{
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
- struct i2c_client *i2c = container_of(codec->dev, struct i2c_client,
- dev);
int i;
- if (i2c->irq)
- free_irq(i2c->irq, codec);
+ if (wm8962->irq)
+ free_irq(wm8962->irq, codec);
cancel_delayed_work_sync(&wm8962->mic_work);
@@ -3986,6 +4034,8 @@
i2c_set_clientdata(i2c, wm8962);
+ wm8962->irq = i2c->irq;
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm8962, &wm8962_dai, 1);
if (ret < 0)
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 9e5ff78..03af7ad 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1434,6 +1434,7 @@
wm8993->hubs_data.hp_startup_mode = 1;
wm8993->hubs_data.dcs_codes = -2;
+ wm8993->hubs_data.series_startup = 1;
ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
if (ret != 0) {
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 970a95c..dc2350e 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2903,6 +2903,7 @@
wm8994->hubs.dcs_codes = -5;
wm8994->hubs.hp_startup_mode = 1;
wm8994->hubs.dcs_readback_mode = 1;
+ wm8994->hubs.series_startup = 1;
break;
default:
wm8994->hubs.dcs_readback_mode = 1;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 91c6b39..a469132 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -727,7 +727,7 @@
SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
+SND_SOC_DAPM_OUT_DRV("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
SND_SOC_DAPM_OUTPUT("SPKN"),
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 9e370d1..2d6c88b 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -107,8 +107,7 @@
return;
}
- /* Devices not using a DCS code correction have startup mode */
- if (hubs->dcs_codes) {
+ if (hubs->series_startup) {
/* Set for 32 series updates */
snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
WM8993_DCS_SERIES_NO_01_MASK,
@@ -134,9 +133,9 @@
break;
case 1:
reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
- reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
+ reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
- reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
+ reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
break;
default:
WARN(1, "Unknown DCS readback method\n");
@@ -150,13 +149,13 @@
dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
hubs->dcs_codes);
- /* HPOUT1L */
- offset = reg_l;
+ /* HPOUT1R */
+ offset = reg_r;
offset += hubs->dcs_codes;
dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
- /* HPOUT1R */
- offset = reg_r;
+ /* HPOUT1L */
+ offset = reg_l;
offset += hubs->dcs_codes;
dcs_cfg |= (u8)offset;
@@ -168,8 +167,8 @@
WM8993_DCS_TRIG_DAC_WR_0 |
WM8993_DCS_TRIG_DAC_WR_1);
} else {
- dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
- dcs_cfg |= reg_r;
+ dcs_cfg = reg_r << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+ dcs_cfg |= reg_l;
}
/* Save the callibrated offset if we're in class W mode and
@@ -195,7 +194,7 @@
/* If we're applying an offset correction then updating the
* callibration would be likely to introduce further offsets. */
- if (hubs->dcs_codes)
+ if (hubs->dcs_codes || hubs->no_series_update)
return ret;
/* Only need to do this if the outputs are active */
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index f8a5e97..0d290d2 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -23,6 +23,8 @@
int dcs_codes;
int dcs_readback_mode;
int hp_startup_mode;
+ int series_startup;
+ int no_series_update;
bool class_w;
u16 class_w_dcs;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 9d35b8c..a49e667 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -46,11 +46,28 @@
}
#endif
+#define DAVINCI_PCM_FMTBITS (\
+ 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_S24_LE |\
+ SNDRV_PCM_FMTBIT_S24_BE |\
+ SNDRV_PCM_FMTBIT_U24_LE |\
+ SNDRV_PCM_FMTBIT_U24_BE |\
+ SNDRV_PCM_FMTBIT_S32_LE |\
+ SNDRV_PCM_FMTBIT_S32_BE |\
+ SNDRV_PCM_FMTBIT_U32_LE |\
+ SNDRV_PCM_FMTBIT_U32_BE)
+
static struct snd_pcm_hardware pcm_hardware_playback = {
.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
- .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME|
+ SNDRV_PCM_INFO_BATCH),
+ .formats = DAVINCI_PCM_FMTBITS,
.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
@@ -59,7 +76,7 @@
.rate_min = 8000,
.rate_max = 96000,
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 384,
.buffer_bytes_max = 128 * 1024,
.period_bytes_min = 32,
.period_bytes_max = 8 * 1024,
@@ -71,8 +88,9 @@
static struct snd_pcm_hardware pcm_hardware_capture = {
.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE),
- .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_BATCH),
+ .formats = DAVINCI_PCM_FMTBITS,
.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
@@ -81,7 +99,7 @@
.rate_min = 8000,
.rate_max = 96000,
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 384,
.buffer_bytes_max = 128 * 1024,
.period_bytes_min = 32,
.period_bytes_max = 8 * 1024,
@@ -139,6 +157,22 @@
struct edmacc_param ram_params;
};
+static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ prtd->period++;
+ if (unlikely(prtd->period >= runtime->periods))
+ prtd->period = 0;
+}
+
+static void davinci_pcm_period_reset(struct snd_pcm_substream *substream)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+
+ prtd->period = 0;
+}
/*
* Not used with ping/pong
*/
@@ -199,10 +233,6 @@
else
edma_set_transfer_params(link, acnt, fifo_level, count,
fifo_level, ABSYNC);
-
- prtd->period++;
- if (unlikely(prtd->period >= runtime->periods))
- prtd->period = 0;
}
static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data)
@@ -217,12 +247,13 @@
return;
if (snd_pcm_running(substream)) {
+ spin_lock(&prtd->lock);
if (prtd->ram_channel < 0) {
/* No ping/pong must fix up link dma data*/
- spin_lock(&prtd->lock);
davinci_pcm_enqueue_dma(substream);
- spin_unlock(&prtd->lock);
}
+ davinci_pcm_period_elapsed(substream);
+ spin_unlock(&prtd->lock);
snd_pcm_period_elapsed(substream);
}
}
@@ -425,7 +456,8 @@
edma_read_slot(link, &prtd->asp_params);
prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN);
- prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f);
+ prtd->asp_params.opt |= TCCHEN |
+ EDMA_TCC(prtd->ram_channel & 0x3f);
edma_write_slot(link, &prtd->asp_params);
/* pong */
@@ -439,7 +471,7 @@
prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f));
/* interrupt after every pong completion */
prtd->asp_params.opt |= TCINTEN | TCCHEN |
- EDMA_TCC(EDMA_CHAN_SLOT(prtd->ram_channel));
+ EDMA_TCC(prtd->ram_channel & 0x3f);
edma_write_slot(link, &prtd->asp_params);
/* ram */
@@ -527,6 +559,13 @@
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
+ edma_start(prtd->asp_channel);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ prtd->ram_channel >= 0) {
+ /* copy 1st iram buffer */
+ edma_start(prtd->ram_channel);
+ }
+ break;
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
edma_resume(prtd->asp_channel);
@@ -550,6 +589,7 @@
{
struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ davinci_pcm_period_reset(substream);
if (prtd->ram_channel >= 0) {
int ret = ping_pong_dma_setup(substream);
if (ret < 0)
@@ -565,21 +605,31 @@
print_buf_info(prtd->asp_link[0], "asp_link[0]");
print_buf_info(prtd->asp_link[1], "asp_link[1]");
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* copy 1st iram buffer */
- edma_start(prtd->ram_channel);
- }
- edma_start(prtd->asp_channel);
+ /*
+ * There is a phase offset of 2 periods between the position
+ * used by dma setup and the position reported in the pointer
+ * function.
+ *
+ * The phase offset, when not using ping-pong buffers, is due to
+ * the two consecutive calls to davinci_pcm_enqueue_dma() below.
+ *
+ * Whereas here, with ping-pong buffers, the phase is due to
+ * there being an entire buffer transfer complete before the
+ * first dma completion event triggers davinci_pcm_dma_irq().
+ */
+ davinci_pcm_period_elapsed(substream);
+ davinci_pcm_period_elapsed(substream);
+
return 0;
}
- prtd->period = 0;
davinci_pcm_enqueue_dma(substream);
+ davinci_pcm_period_elapsed(substream);
/* Copy self-linked parameter RAM entry into master channel */
edma_read_slot(prtd->asp_link[0], &prtd->asp_params);
edma_write_slot(prtd->asp_channel, &prtd->asp_params);
davinci_pcm_enqueue_dma(substream);
- edma_start(prtd->asp_channel);
+ davinci_pcm_period_elapsed(substream);
return 0;
}
@@ -591,51 +641,23 @@
struct davinci_runtime_data *prtd = runtime->private_data;
unsigned int offset;
int asp_count;
- dma_addr_t asp_src, asp_dst;
+ unsigned int period_size = snd_pcm_lib_period_bytes(substream);
+ /*
+ * There is a phase offset of 2 periods between the position used by dma
+ * setup and the position reported in the pointer function. Either +2 in
+ * the dma setup or -2 here in the pointer function (with wrapping,
+ * both) accounts for this offset -- choose the latter since it makes
+ * the first-time setup clearer.
+ */
spin_lock(&prtd->lock);
- if (prtd->ram_channel >= 0) {
- int ram_count;
- int mod_ram;
- dma_addr_t ram_src, ram_dst;
- unsigned int period_size = snd_pcm_lib_period_bytes(substream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* reading ram before asp should be safe
- * as long as the asp transfers less than a ping size
- * of bytes between the 2 reads
- */
- edma_get_position(prtd->ram_channel,
- &ram_src, &ram_dst);
- edma_get_position(prtd->asp_channel,
- &asp_src, &asp_dst);
- asp_count = asp_src - prtd->asp_params.src;
- ram_count = ram_src - prtd->ram_params.src;
- mod_ram = ram_count % period_size;
- mod_ram -= asp_count;
- if (mod_ram < 0)
- mod_ram += period_size;
- else if (mod_ram == 0) {
- if (snd_pcm_running(substream))
- mod_ram += period_size;
- }
- ram_count -= mod_ram;
- if (ram_count < 0)
- ram_count += period_size * runtime->periods;
- } else {
- edma_get_position(prtd->ram_channel,
- &ram_src, &ram_dst);
- ram_count = ram_dst - prtd->ram_params.dst;
- }
- asp_count = ram_count;
- } else {
- edma_get_position(prtd->asp_channel, &asp_src, &asp_dst);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- asp_count = asp_src - runtime->dma_addr;
- else
- asp_count = asp_dst - runtime->dma_addr;
- }
+ asp_count = prtd->period - 2;
spin_unlock(&prtd->lock);
+ if (asp_count < 0)
+ asp_count += runtime->periods;
+ asp_count *= period_size;
+
offset = bytes_to_frames(runtime, asp_count);
if (offset >= runtime->buffer_size)
offset = 0;
@@ -811,9 +833,11 @@
static u64 davinci_pcm_dmamask = 0xffffffff;
-static int davinci_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret;
if (!card->dev->dma_mask)
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index a456e49..e27c417 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -266,9 +266,11 @@
static u64 ep93xx_pcm_dmamask = 0xffffffff;
-static int ep93xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 6680c0b..732208c 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -294,9 +294,11 @@
* Regardless of where the memory is actually allocated, since the device can
* technically DMA to any 36-bit address, we do need to set the DMA mask to 36.
*/
-static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
static u64 fsl_dma_dmamask = DMA_BIT_MASK(36);
int ret;
@@ -939,7 +941,7 @@
iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL);
if (iprop)
- dma->ssi_fifo_depth = *iprop;
+ dma->ssi_fifo_depth = be32_to_cpup(iprop);
else
/* Older 8610 DTs didn't have the fifo-depth property */
dma->ssi_fifo_depth = 8;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 313e0cc..d48afea 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -678,7 +678,12 @@
kfree(ssi_private);
return ret;
}
- ssi_private->ssi = ioremap(res.start, 1 + res.end - res.start);
+ ssi_private->ssi = of_iomap(np, 0);
+ if (!ssi_private->ssi) {
+ dev_err(&pdev->dev, "could not map device resources\n");
+ kfree(ssi_private);
+ return -ENOMEM;
+ }
ssi_private->ssi_phys = res.start;
ssi_private->irq = irq_of_parse_and_map(np, 0);
@@ -691,7 +696,7 @@
/* Determine the FIFO depth. */
iprop = of_get_property(np, "fsl,fifo-depth", NULL);
if (iprop)
- ssi_private->fifo_depth = *iprop;
+ ssi_private->fifo_depth = be32_to_cpup(iprop);
else
/* Older 8610 DTs didn't have the fifo-depth property */
ssi_private->fifo_depth = 8;
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index fff695c..19ad0c1 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -299,10 +299,11 @@
};
static u64 psc_dma_dmamask = 0xffffffff;
-static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int psc_dma_new(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
size_t size = psc_dma_hardware.buffer_bytes_max;
int rc = 0;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index c16c6b2..a192979 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -233,7 +233,7 @@
if (!iprop)
return -1;
- return *iprop;
+ return be32_to_cpup(iprop);
}
/**
@@ -258,7 +258,7 @@
if (!iprop)
return -EINVAL;
- addr = *iprop;
+ addr = be32_to_cpup(iprop);
bus = get_parent_cell_index(np);
if (bus < 0)
@@ -305,7 +305,7 @@
return -EINVAL;
}
- *dma_channel_id = *iprop;
+ *dma_channel_id = be32_to_cpup(iprop);
*dma_id = get_parent_cell_index(dma_channel_np);
of_node_put(dma_channel_np);
@@ -379,7 +379,7 @@
ret = -EINVAL;
goto error;
}
- machine_data->ssi_id = *iprop;
+ machine_data->ssi_id = be32_to_cpup(iprop);
/* Get the serial format and clock direction. */
sprop = of_get_property(np, "fsl,mode", NULL);
@@ -405,7 +405,7 @@
ret = -EINVAL;
goto error;
}
- machine_data->clk_frequency = *iprop;
+ machine_data->clk_frequency = be32_to_cpup(iprop);
} else if (strcasecmp(sprop, "i2s-master") == 0) {
machine_data->dai_format = SND_SOC_DAIFMT_I2S;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 66e0b68..8fa4d5f 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -232,7 +232,7 @@
iprop = of_get_property(parent, "cell-index", NULL);
if (iprop)
- ret = *iprop;
+ ret = be32_to_cpup(iprop);
of_node_put(parent);
@@ -261,7 +261,7 @@
if (!iprop)
return -EINVAL;
- addr = *iprop;
+ addr = be32_to_cpup(iprop);
bus = get_parent_cell_index(np);
if (bus < 0)
@@ -308,7 +308,7 @@
return -EINVAL;
}
- *dma_channel_id = *iprop;
+ *dma_channel_id = be32_to_cpup(iprop);
*dma_id = get_parent_cell_index(dma_channel_np);
of_node_put(dma_channel_np);
@@ -379,7 +379,7 @@
ret = -EINVAL;
goto error;
}
- mdata->ssi_id = *iprop;
+ mdata->ssi_id = be32_to_cpup(iprop);
/* Get the serial format and clock direction. */
sprop = of_get_property(np, "fsl,mode", NULL);
@@ -405,7 +405,7 @@
ret = -EINVAL;
goto error;
}
- mdata->clk_frequency = *iprop;
+ mdata->clk_frequency = be32_to_cpup(iprop);
} else if (strcasecmp(sprop, "i2s-master") == 0) {
mdata->dai_format = SND_SOC_DAIFMT_I2S;
mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 413b78d..309c59e 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -238,12 +238,14 @@
static int ssi_irq = 0;
-static int imx_pcm_fiq_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret;
- ret = imx_pcm_new(card, dai, pcm);
+ ret = imx_pcm_new(rtd);
if (ret)
return ret;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 61fceb0..10a8e27 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -388,10 +388,11 @@
static u64 imx_pcm_dmamask = DMA_BIT_MASK(32);
-int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+int imx_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
-
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h
index dc8a875..0a84cec 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/imx/imx-ssi.h
@@ -225,8 +225,7 @@
struct imx_ssi *ssi);
int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma);
-int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm);
+int imx_pcm_new(struct snd_soc_pcm_runtime *rtd);
void imx_pcm_free(struct snd_pcm *pcm);
/*
diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c
index fb1483f..a7c9578 100644
--- a/sound/soc/jz4740/jz4740-pcm.c
+++ b/sound/soc/jz4740/jz4740-pcm.c
@@ -299,9 +299,11 @@
static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32);
-int jz4740_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index e13c6ce..cd33de1 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -312,9 +312,11 @@
return 0;
}
-static int kirkwood_dma_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret;
if (!card->dev->dma_mask)
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index 5a946b4..3e78260 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -402,9 +402,10 @@
snd_pcm_lib_preallocate_free_for_all(pcm);
}
-int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int retval = 0;
pr_debug("sst_pcm_new called\n");
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index dac6732..9c0edad 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -356,7 +356,7 @@
nuc900_audio->irq_num = platform_get_irq(pdev, 0);
if (!nuc900_audio->irq_num) {
ret = -EBUSY;
- goto out2;
+ goto out3;
}
nuc900_ac97_data = nuc900_audio;
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index 8263f56..d589ef1 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -315,9 +315,12 @@
}
static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32);
-static int nuc900_dma_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
+
if (!card->dev->dma_mask)
card->dev->dma_mask = &nuc900_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 462cbcb..b40095a 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -427,7 +427,8 @@
/* Board specific codec bias level control */
static int ams_delta_set_bias_level(struct snd_soc_card *card,
- enum snd_soc_bias_level level)
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
{
struct snd_soc_codec *codec = card->rtd->codec;
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index e6a6b99..b2f5751 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -366,9 +366,11 @@
}
}
-static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index fab20a5..da28394 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -85,9 +85,11 @@
static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32);
-static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index ab3ccae..80c85fd6 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -443,10 +443,11 @@
static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32);
-static int s6000_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime)
{
- struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+ struct snd_card *card = runtime->card->snd_card;
+ struct snd_soc_dai *dai = runtime->cpu_dai;
+ struct snd_pcm *pcm = runtime->pcm;
struct s6000_pcm_dma_params *params;
int res;
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index d155cbb..54b0e4b 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -158,7 +158,7 @@
config SND_SOC_SAMSUNG_SMDK_SPDIF
tristate "SoC S/PDIF Audio support for SMDK"
- depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210)
+ depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310)
select SND_SAMSUNG_SPDIF
help
Say Y if you want to add support for SoC S/PDIF audio on the SMDK.
@@ -171,9 +171,23 @@
help
Say Y if you want to add support for SoC audio on the SMDK.
+config SND_SOC_SMDK_WM8994_PCM
+ tristate "SoC PCM Audio support for WM8994 on SMDK"
+ depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310)
+ select SND_SOC_WM8994
+ select SND_SAMSUNG_PCM
+ help
+ Say Y if you want to add support for SoC audio on the SMDK
+
config SND_SOC_SPEYSIDE
tristate "Audio support for Wolfson Speyside"
depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
select SND_SAMSUNG_I2S
select SND_SOC_WM8915
select SND_SOC_WM9081
+
+config SND_SOC_SPEYSIDE_WM8962
+ tristate "Audio support for Wolfson Speyside with WM8962"
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ select SND_SAMSUNG_I2S
+ select SND_SOC_WM8962
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 683843a..9eb3b12 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -35,7 +35,9 @@
snd-soc-goni-wm8994-objs := goni_wm8994.o
snd-soc-smdk-spdif-objs := smdk_spdif.o
snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o
+snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o
snd-soc-speyside-objs := speyside.o
+snd-soc-speyside-wm8962-objs := speyside_wm8962.o
obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -54,4 +56,6 @@
obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o
obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o
obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o
+obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o
obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o
+obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 5cb3b88..9465588 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -425,9 +425,11 @@
static u64 dma_mask = DMA_BIT_MASK(32);
-static int dma_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int dma_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
pr_debug("Entered %s\n", __func__);
diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h
new file mode 100644
index 0000000..c0e6d9a
--- /dev/null
+++ b/sound/soc/samsung/i2s-regs.h
@@ -0,0 +1,143 @@
+/*
+ * linux/sound/soc/samsung/i2s-regs.h
+ *
+ * Copyright (c) 2011 Samsung Electronics Co., Ltd.
+ * http://www.samsung.com
+ *
+ * Samsung I2S driver's register header
+ *
+ * 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.
+ */
+
+#ifndef __SND_SOC_SAMSUNG_I2S_REGS_H
+#define __SND_SOC_SAMSUNG_I2S_REGS_H
+
+#define I2SCON 0x0
+#define I2SMOD 0x4
+#define I2SFIC 0x8
+#define I2SPSR 0xc
+#define I2STXD 0x10
+#define I2SRXD 0x14
+#define I2SFICS 0x18
+#define I2STXDS 0x1c
+#define I2SAHB 0x20
+#define I2SSTR0 0x24
+#define I2SSIZE 0x28
+#define I2STRNCNT 0x2c
+#define I2SLVL0ADDR 0x30
+#define I2SLVL1ADDR 0x34
+#define I2SLVL2ADDR 0x38
+#define I2SLVL3ADDR 0x3c
+
+#define CON_RSTCLR (1 << 31)
+#define CON_FRXOFSTATUS (1 << 26)
+#define CON_FRXORINTEN (1 << 25)
+#define CON_FTXSURSTAT (1 << 24)
+#define CON_FTXSURINTEN (1 << 23)
+#define CON_TXSDMA_PAUSE (1 << 20)
+#define CON_TXSDMA_ACTIVE (1 << 18)
+
+#define CON_FTXURSTATUS (1 << 17)
+#define CON_FTXURINTEN (1 << 16)
+#define CON_TXFIFO2_EMPTY (1 << 15)
+#define CON_TXFIFO1_EMPTY (1 << 14)
+#define CON_TXFIFO2_FULL (1 << 13)
+#define CON_TXFIFO1_FULL (1 << 12)
+
+#define CON_LRINDEX (1 << 11)
+#define CON_TXFIFO_EMPTY (1 << 10)
+#define CON_RXFIFO_EMPTY (1 << 9)
+#define CON_TXFIFO_FULL (1 << 8)
+#define CON_RXFIFO_FULL (1 << 7)
+#define CON_TXDMA_PAUSE (1 << 6)
+#define CON_RXDMA_PAUSE (1 << 5)
+#define CON_TXCH_PAUSE (1 << 4)
+#define CON_RXCH_PAUSE (1 << 3)
+#define CON_TXDMA_ACTIVE (1 << 2)
+#define CON_RXDMA_ACTIVE (1 << 1)
+#define CON_ACTIVE (1 << 0)
+
+#define MOD_OPCLK_CDCLK_OUT (0 << 30)
+#define MOD_OPCLK_CDCLK_IN (1 << 30)
+#define MOD_OPCLK_BCLK_OUT (2 << 30)
+#define MOD_OPCLK_PCLK (3 << 30)
+#define MOD_OPCLK_MASK (3 << 30)
+#define MOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */
+
+#define MOD_BLCS_SHIFT 26
+#define MOD_BLCS_16BIT (0 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_8BIT (1 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_24BIT (2 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_MASK (3 << MOD_BLCS_SHIFT)
+#define MOD_BLCP_SHIFT 24
+#define MOD_BLCP_16BIT (0 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_8BIT (1 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_24BIT (2 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_MASK (3 << MOD_BLCP_SHIFT)
+
+#define MOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */
+#define MOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */
+#define MOD_C1DD_HHALF (1 << 19)
+#define MOD_C1DD_LHALF (1 << 18)
+#define MOD_DC2_EN (1 << 17)
+#define MOD_DC1_EN (1 << 16)
+#define MOD_BLC_16BIT (0 << 13)
+#define MOD_BLC_8BIT (1 << 13)
+#define MOD_BLC_24BIT (2 << 13)
+#define MOD_BLC_MASK (3 << 13)
+
+#define MOD_IMS_SYSMUX (1 << 10)
+#define MOD_SLAVE (1 << 11)
+#define MOD_TXONLY (0 << 8)
+#define MOD_RXONLY (1 << 8)
+#define MOD_TXRX (2 << 8)
+#define MOD_MASK (3 << 8)
+#define MOD_LR_LLOW (0 << 7)
+#define MOD_LR_RLOW (1 << 7)
+#define MOD_SDF_IIS (0 << 5)
+#define MOD_SDF_MSB (1 << 5)
+#define MOD_SDF_LSB (2 << 5)
+#define MOD_SDF_MASK (3 << 5)
+#define MOD_RCLK_256FS (0 << 3)
+#define MOD_RCLK_512FS (1 << 3)
+#define MOD_RCLK_384FS (2 << 3)
+#define MOD_RCLK_768FS (3 << 3)
+#define MOD_RCLK_MASK (3 << 3)
+#define MOD_BCLK_32FS (0 << 1)
+#define MOD_BCLK_48FS (1 << 1)
+#define MOD_BCLK_16FS (2 << 1)
+#define MOD_BCLK_24FS (3 << 1)
+#define MOD_BCLK_MASK (3 << 1)
+#define MOD_8BIT (1 << 0)
+
+#define MOD_CDCLKCON (1 << 12)
+
+#define PSR_PSREN (1 << 15)
+
+#define FIC_TX2COUNT(x) (((x) >> 24) & 0xf)
+#define FIC_TX1COUNT(x) (((x) >> 16) & 0xf)
+
+#define FIC_TXFLUSH (1 << 15)
+#define FIC_RXFLUSH (1 << 7)
+
+#define FIC_TXCOUNT(x) (((x) >> 8) & 0xf)
+#define FIC_RXCOUNT(x) (((x) >> 0) & 0xf)
+#define FICS_TXCOUNT(x) (((x) >> 8) & 0x7f)
+
+#define AHB_INTENLVL0 (1 << 24)
+#define AHB_LVL0INT (1 << 20)
+#define AHB_CLRLVL0INT (1 << 16)
+#define AHB_DMARLD (1 << 5)
+#define AHB_INTMASK (1 << 3)
+#define AHB_DMAEN (1 << 0)
+#define AHB_LVLINTMASK (0xf << 20)
+
+#define I2SSIZE_TRNMSK (0xffff)
+#define I2SSIZE_SHIFT (16)
+
+#endif /* __SND_SOC_SAMSUNG_I2S_REGS_H */
+
+
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 992a732..1568eea 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -22,109 +22,7 @@
#include "dma.h"
#include "i2s.h"
-
-#define I2SCON 0x0
-#define I2SMOD 0x4
-#define I2SFIC 0x8
-#define I2SPSR 0xc
-#define I2STXD 0x10
-#define I2SRXD 0x14
-#define I2SFICS 0x18
-#define I2STXDS 0x1c
-
-#define CON_RSTCLR (1 << 31)
-#define CON_FRXOFSTATUS (1 << 26)
-#define CON_FRXORINTEN (1 << 25)
-#define CON_FTXSURSTAT (1 << 24)
-#define CON_FTXSURINTEN (1 << 23)
-#define CON_TXSDMA_PAUSE (1 << 20)
-#define CON_TXSDMA_ACTIVE (1 << 18)
-
-#define CON_FTXURSTATUS (1 << 17)
-#define CON_FTXURINTEN (1 << 16)
-#define CON_TXFIFO2_EMPTY (1 << 15)
-#define CON_TXFIFO1_EMPTY (1 << 14)
-#define CON_TXFIFO2_FULL (1 << 13)
-#define CON_TXFIFO1_FULL (1 << 12)
-
-#define CON_LRINDEX (1 << 11)
-#define CON_TXFIFO_EMPTY (1 << 10)
-#define CON_RXFIFO_EMPTY (1 << 9)
-#define CON_TXFIFO_FULL (1 << 8)
-#define CON_RXFIFO_FULL (1 << 7)
-#define CON_TXDMA_PAUSE (1 << 6)
-#define CON_RXDMA_PAUSE (1 << 5)
-#define CON_TXCH_PAUSE (1 << 4)
-#define CON_RXCH_PAUSE (1 << 3)
-#define CON_TXDMA_ACTIVE (1 << 2)
-#define CON_RXDMA_ACTIVE (1 << 1)
-#define CON_ACTIVE (1 << 0)
-
-#define MOD_OPCLK_CDCLK_OUT (0 << 30)
-#define MOD_OPCLK_CDCLK_IN (1 << 30)
-#define MOD_OPCLK_BCLK_OUT (2 << 30)
-#define MOD_OPCLK_PCLK (3 << 30)
-#define MOD_OPCLK_MASK (3 << 30)
-#define MOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */
-
-#define MOD_BLCS_SHIFT 26
-#define MOD_BLCS_16BIT (0 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_8BIT (1 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_24BIT (2 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_MASK (3 << MOD_BLCS_SHIFT)
-#define MOD_BLCP_SHIFT 24
-#define MOD_BLCP_16BIT (0 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_8BIT (1 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_24BIT (2 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_MASK (3 << MOD_BLCP_SHIFT)
-
-#define MOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */
-#define MOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */
-#define MOD_C1DD_HHALF (1 << 19)
-#define MOD_C1DD_LHALF (1 << 18)
-#define MOD_DC2_EN (1 << 17)
-#define MOD_DC1_EN (1 << 16)
-#define MOD_BLC_16BIT (0 << 13)
-#define MOD_BLC_8BIT (1 << 13)
-#define MOD_BLC_24BIT (2 << 13)
-#define MOD_BLC_MASK (3 << 13)
-
-#define MOD_IMS_SYSMUX (1 << 10)
-#define MOD_SLAVE (1 << 11)
-#define MOD_TXONLY (0 << 8)
-#define MOD_RXONLY (1 << 8)
-#define MOD_TXRX (2 << 8)
-#define MOD_MASK (3 << 8)
-#define MOD_LR_LLOW (0 << 7)
-#define MOD_LR_RLOW (1 << 7)
-#define MOD_SDF_IIS (0 << 5)
-#define MOD_SDF_MSB (1 << 5)
-#define MOD_SDF_LSB (2 << 5)
-#define MOD_SDF_MASK (3 << 5)
-#define MOD_RCLK_256FS (0 << 3)
-#define MOD_RCLK_512FS (1 << 3)
-#define MOD_RCLK_384FS (2 << 3)
-#define MOD_RCLK_768FS (3 << 3)
-#define MOD_RCLK_MASK (3 << 3)
-#define MOD_BCLK_32FS (0 << 1)
-#define MOD_BCLK_48FS (1 << 1)
-#define MOD_BCLK_16FS (2 << 1)
-#define MOD_BCLK_24FS (3 << 1)
-#define MOD_BCLK_MASK (3 << 1)
-#define MOD_8BIT (1 << 0)
-
-#define MOD_CDCLKCON (1 << 12)
-
-#define PSR_PSREN (1 << 15)
-
-#define FIC_TX2COUNT(x) (((x) >> 24) & 0xf)
-#define FIC_TX1COUNT(x) (((x) >> 16) & 0xf)
-
-#define FIC_TXFLUSH (1 << 15)
-#define FIC_RXFLUSH (1 << 7)
-#define FIC_TXCOUNT(x) (((x) >> 8) & 0xf)
-#define FIC_RXCOUNT(x) (((x) >> 0) & 0xf)
-#define FICS_TXCOUNT(x) (((x) >> 8) & 0x7f)
+#include "i2s-regs.h"
#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t)
diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c
new file mode 100644
index 0000000..5f21116
--- /dev/null
+++ b/sound/soc/samsung/smdk_wm8994pcm.c
@@ -0,0 +1,176 @@
+/*
+ * sound/soc/samsung/smdk_wm8994pcm.c
+ *
+ * Copyright (c) 2011 Samsung Electronics Co., Ltd
+ * http://www.samsung.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 <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/wm8994.h"
+#include "dma.h"
+#include "pcm.h"
+
+/*
+ * Board Settings:
+ * o '1' means 'ON'
+ * o '0' means 'OFF'
+ * o 'X' means 'Don't care'
+ *
+ * SMDKC210, SMDKV310: CFG3- 1001, CFG5-1000, CFG7-111111
+ */
+
+/*
+ * Configure audio route as :-
+ * $ amixer sset 'DAC1' on,on
+ * $ amixer sset 'Right Headphone Mux' 'DAC'
+ * $ amixer sset 'Left Headphone Mux' 'DAC'
+ * $ amixer sset 'DAC1R Mixer AIF1.1' on
+ * $ amixer sset 'DAC1L Mixer AIF1.1' on
+ * $ amixer sset 'IN2L' on
+ * $ amixer sset 'IN2L PGA IN2LN' on
+ * $ amixer sset 'MIXINL IN2L' on
+ * $ amixer sset 'AIF1ADC1L Mixer ADC/DMIC' on
+ * $ amixer sset 'IN2R' on
+ * $ amixer sset 'IN2R PGA IN2RN' on
+ * $ amixer sset 'MIXINR IN2R' on
+ * $ amixer sset 'AIF1ADC1R Mixer ADC/DMIC' on
+ */
+
+/* SMDK has a 16.9344MHZ crystal attached to WM8994 */
+#define SMDK_WM8994_FREQ 16934400
+
+static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned long mclk_freq;
+ int rfs, ret;
+
+ switch(params_rate(params)) {
+ case 8000:
+ rfs = 512;
+ break;
+ default:
+ dev_err(cpu_dai->dev, "%s:%d Sampling Rate %u not supported!\n",
+ __func__, __LINE__, params_rate(params));
+ return -EINVAL;
+ }
+
+ mclk_freq = params_rate(params) * rfs;
+
+ /* Set the codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B
+ | SND_SOC_DAIFMT_IB_NF
+ | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* Set the cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B
+ | SND_SOC_DAIFMT_IB_NF
+ | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1,
+ mclk_freq, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1,
+ SMDK_WM8994_FREQ, mclk_freq);
+ if (ret < 0)
+ return ret;
+
+ /* Set PCM source clock on CPU */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX,
+ mclk_freq, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* Set SCLK_DIV for making bclk */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops smdk_wm8994_pcm_ops = {
+ .hw_params = smdk_wm8994_pcm_hw_params,
+};
+
+static struct snd_soc_dai_link smdk_dai[] = {
+ {
+ .name = "WM8994 PAIF PCM",
+ .stream_name = "Primary PCM",
+ .cpu_dai_name = "samsung-pcm.0",
+ .codec_dai_name = "wm8994-aif1",
+ .platform_name = "samsung-audio",
+ .codec_name = "wm8994-codec",
+ .ops = &smdk_wm8994_pcm_ops,
+ },
+};
+
+static struct snd_soc_card smdk_pcm = {
+ .name = "SMDK-PCM",
+ .dai_link = smdk_dai,
+ .num_links = 1,
+};
+
+static int __devinit snd_smdk_probe(struct platform_device *pdev)
+{
+ int ret = 0;
+
+ smdk_pcm.dev = &pdev->dev;
+ ret = snd_soc_register_card(&smdk_pcm);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int __devexit snd_smdk_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&smdk_pcm);
+ platform_set_drvdata(pdev, NULL);
+ return 0;
+}
+
+static struct platform_driver snd_smdk_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "samsung-smdk-pcm",
+ },
+ .probe = snd_smdk_probe,
+ .remove = __devexit_p(snd_smdk_remove),
+};
+
+static int __init smdk_audio_init(void)
+{
+ return platform_driver_register(&snd_smdk_driver);
+}
+
+module_init(smdk_audio_init);
+
+static void __exit smdk_audio_exit(void)
+{
+ platform_driver_unregister(&snd_smdk_driver);
+}
+
+module_exit(smdk_audio_exit);
+
+MODULE_AUTHOR("Sangbeom Kim, <sbkim73@samsung.com>");
+MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 360a333..d6dee4d 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -20,24 +20,29 @@
#define WM8915_HPSEL_GPIO 214
static int speyside_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
int ret;
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
switch (level) {
case SND_SOC_BIAS_STANDBY:
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK1,
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK2,
32768, SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK1,
+ ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK2,
0, 0, 0);
if (ret < 0) {
pr_err("Failed to stop FLL\n");
return ret;
}
+ break;
default:
break;
@@ -46,6 +51,45 @@
return 0;
}
+static int speyside_set_bias_level_post(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ int ret;
+
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
+ ret = snd_soc_dai_set_pll(codec_dai, 0,
+ WM8915_FLL_MCLK2,
+ 32768, 48000 * 256);
+ if (ret < 0) {
+ pr_err("Failed to start FLL\n");
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ WM8915_SYSCLK_FLL,
+ 48000 * 256,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ card->dapm.bias_level = level;
+
+ return 0;
+}
+
static int speyside_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -66,16 +110,6 @@
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_pll(codec_dai, 0, WM8915_FLL_MCLK1,
- 32768, 256 * 48000);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_FLL,
- 256 * 48000, SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
return 0;
}
@@ -127,7 +161,7 @@
struct snd_soc_codec *codec = rtd->codec;
int ret;
- ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK1, 32768, 0);
+ ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK2, 32768, 0);
if (ret < 0)
return ret;
@@ -267,6 +301,7 @@
.num_configs = ARRAY_SIZE(speyside_codec_conf),
.set_bias_level = speyside_set_bias_level,
+ .set_bias_level_post = speyside_set_bias_level_post,
.controls = controls,
.num_controls = ARRAY_SIZE(controls),
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
new file mode 100644
index 0000000..c0ba0bf
--- /dev/null
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -0,0 +1,260 @@
+/*
+ * Speyside with WM8962 audio support
+ *
+ * Copyright 2011 Wolfson Microelectronics
+ *
+ * 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 <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+#include <linux/gpio.h>
+
+#include "../codecs/wm8962.h"
+
+static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ WM8962_FLL_MCLK, 32768,
+ 44100 * 256);
+ if (ret < 0)
+ pr_err("Failed to start FLL\n");
+
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ WM8962_SYSCLK_FLL,
+ 44100 * 256,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_STANDBY:
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+ 32768, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ 0, 0, 0);
+ if (ret < 0) {
+ pr_err("Failed to stop FLL\n");
+ return ret;
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ dapm->bias_level = level;
+
+ return 0;
+}
+
+static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops speyside_wm8962_ops = {
+ .hw_params = speyside_wm8962_hw_params,
+};
+
+static struct snd_soc_dai_link speyside_wm8962_dai[] = {
+ {
+ .name = "CPU",
+ .stream_name = "CPU",
+ .cpu_dai_name = "samsung-i2s.0",
+ .codec_dai_name = "wm8962",
+ .platform_name = "samsung-audio",
+ .codec_name = "wm8962.1-001a",
+ .ops = &speyside_wm8962_ops,
+ },
+};
+
+static const struct snd_kcontrol_new controls[] = {
+ SOC_DAPM_PIN_SWITCH("Main Speaker"),
+};
+
+static struct snd_soc_dapm_widget widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+
+ SND_SOC_DAPM_MIC("DMIC", NULL),
+
+ SND_SOC_DAPM_SPK("Main Speaker", NULL),
+};
+
+static struct snd_soc_dapm_route audio_paths[] = {
+ { "Headphone", NULL, "HPOUTL" },
+ { "Headphone", NULL, "HPOUTR" },
+
+ { "Main Speaker", NULL, "SPKOUTL" },
+ { "Main Speaker", NULL, "SPKOUTR" },
+
+ { "MICBIAS", NULL, "Headset Mic" },
+ { "IN4L", NULL, "MICBIAS" },
+ { "IN4R", NULL, "MICBIAS" },
+
+ { "MICBIAS", NULL, "DMIC" },
+ { "DMICDAT", NULL, "MICBIAS" },
+};
+
+static struct snd_soc_jack speyside_wm8962_headset;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin speyside_wm8962_headset_pins[] = {
+ {
+ .pin = "Headset Mic",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Headphone",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+static int speyside_wm8962_late_probe(struct snd_soc_card *card)
+{
+ struct snd_soc_codec *codec = card->rtd[0].codec;
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+ 32768, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_jack_new(codec, "Headset",
+ SND_JACK_HEADSET | SND_JACK_BTN_0,
+ &speyside_wm8962_headset);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_jack_add_pins(&speyside_wm8962_headset,
+ ARRAY_SIZE(speyside_wm8962_headset_pins),
+ speyside_wm8962_headset_pins);
+ if (ret)
+ return ret;
+
+ wm8962_mic_detect(codec, &speyside_wm8962_headset);
+
+ return 0;
+}
+
+static struct snd_soc_card speyside_wm8962 = {
+ .name = "Speyside WM8962",
+ .dai_link = speyside_wm8962_dai,
+ .num_links = ARRAY_SIZE(speyside_wm8962_dai),
+
+ .set_bias_level = speyside_wm8962_set_bias_level,
+ .set_bias_level_post = speyside_wm8962_set_bias_level_post,
+
+ .controls = controls,
+ .num_controls = ARRAY_SIZE(controls),
+ .dapm_widgets = widgets,
+ .num_dapm_widgets = ARRAY_SIZE(widgets),
+ .dapm_routes = audio_paths,
+ .num_dapm_routes = ARRAY_SIZE(audio_paths),
+
+ .late_probe = speyside_wm8962_late_probe,
+};
+
+static __devinit int speyside_wm8962_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &speyside_wm8962;
+ int ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int __devexit speyside_wm8962_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver speyside_wm8962_driver = {
+ .driver = {
+ .name = "speyside-wm8962",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = speyside_wm8962_probe,
+ .remove = __devexit_p(speyside_wm8962_remove),
+};
+
+static int __init speyside_wm8962_audio_init(void)
+{
+ return platform_driver_register(&speyside_wm8962_driver);
+}
+module_init(speyside_wm8962_audio_init);
+
+static void __exit speyside_wm8962_audio_exit(void)
+{
+ platform_driver_unregister(&speyside_wm8962_driver);
+}
+module_exit(speyside_wm8962_audio_exit);
+
+MODULE_DESCRIPTION("Speyside WM8962 audio support");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:speyside-wm8962");
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index c326d29..db74005 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -327,10 +327,10 @@
snd_pcm_lib_preallocate_free_for_all(pcm);
}
-static int camelot_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_pcm *pcm = rtd->pcm;
+
/* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
* in MMAP mode (i.e. aplay -M)
*/
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 4a9da6b..8e112cc 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -118,10 +118,38 @@
/*
* FSI driver use below type name for variable
*
- * xxx_len : data length
- * xxx_width : data width
- * xxx_offset : data offset
* xxx_num : number of data
+ * xxx_pos : position of data
+ * xxx_capa : capacity of data
+ */
+
+/*
+ * period/frame/sample image
+ *
+ * ex) PCM (2ch)
+ *
+ * period pos period pos
+ * [n] [n + 1]
+ * |<-------------------- period--------------------->|
+ * ==|============================================ ... =|==
+ * | |
+ * ||<----- frame ----->|<------ frame ----->| ... |
+ * |+--------------------+--------------------+- ... |
+ * ||[ sample ][ sample ]|[ sample ][ sample ]| ... |
+ * |+--------------------+--------------------+- ... |
+ * ==|============================================ ... =|==
+ */
+
+/*
+ * FSI FIFO image
+ *
+ * | |
+ * | |
+ * | [ sample ] |
+ * | [ sample ] |
+ * | [ sample ] |
+ * | [ sample ] |
+ * --> go to codecs
*/
/*
@@ -131,12 +159,11 @@
struct fsi_stream {
struct snd_pcm_substream *substream;
- int fifo_max_num;
-
- int buff_offset;
- int buff_len;
- int period_len;
- int period_num;
+ int fifo_sample_capa; /* sample capacity of FSI FIFO */
+ int buff_sample_capa; /* sample capacity of ALSA buffer */
+ int buff_sample_pos; /* sample position of ALSA buffer */
+ int period_samples; /* sample number / 1 period */
+ int period_pos; /* current period position */
int uerr_num;
int oerr_num;
@@ -149,17 +176,14 @@
struct fsi_stream playback;
struct fsi_stream capture;
+ u32 do_fmt;
+ u32 di_fmt;
+
int chan_num:16;
int clk_master:1;
+ int spdif:1;
long rate;
-
- /* for suspend/resume */
- u32 saved_do_fmt;
- u32 saved_di_fmt;
- u32 saved_ckg1;
- u32 saved_ckg2;
- u32 saved_out_sel;
};
struct fsi_core {
@@ -180,14 +204,6 @@
struct fsi_core *core;
struct sh_fsi_platform_info *info;
spinlock_t lock;
-
- /* for suspend/resume */
- u32 saved_a_mclk;
- u32 saved_b_mclk;
- u32 saved_iemsk;
- u32 saved_imsk;
- u32 saved_clk_rst;
- u32 saved_soft_rst;
};
/*
@@ -271,6 +287,11 @@
return fsi->master->base == fsi->base;
}
+static int fsi_is_spdif(struct fsi_priv *fsi)
+{
+ return fsi->spdif;
+}
+
static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -342,28 +363,59 @@
return shift;
}
-static void fsi_stream_push(struct fsi_priv *fsi,
- int is_play,
- struct snd_pcm_substream *substream,
- u32 buffer_len,
- u32 period_len)
+static int fsi_frame2sample(struct fsi_priv *fsi, int frames)
+{
+ return frames * fsi->chan_num;
+}
+
+static int fsi_sample2frame(struct fsi_priv *fsi, int samples)
+{
+ return samples / fsi->chan_num;
+}
+
+static int fsi_stream_is_working(struct fsi_priv *fsi,
+ int is_play)
{
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
+ struct fsi_master *master = fsi_get_master(fsi);
+ unsigned long flags;
+ int ret;
+ spin_lock_irqsave(&master->lock, flags);
+ ret = !!io->substream;
+ spin_unlock_irqrestore(&master->lock, flags);
+
+ return ret;
+}
+
+static void fsi_stream_push(struct fsi_priv *fsi,
+ int is_play,
+ struct snd_pcm_substream *substream)
+{
+ struct fsi_stream *io = fsi_get_stream(fsi, is_play);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsi_master *master = fsi_get_master(fsi);
+ unsigned long flags;
+
+ spin_lock_irqsave(&master->lock, flags);
io->substream = substream;
- io->buff_len = buffer_len;
- io->buff_offset = 0;
- io->period_len = period_len;
- io->period_num = 0;
+ io->buff_sample_capa = fsi_frame2sample(fsi, runtime->buffer_size);
+ io->buff_sample_pos = 0;
+ io->period_samples = fsi_frame2sample(fsi, runtime->period_size);
+ io->period_pos = 0;
io->oerr_num = -1; /* ignore 1st err */
io->uerr_num = -1; /* ignore 1st err */
+ spin_unlock_irqrestore(&master->lock, flags);
}
static void fsi_stream_pop(struct fsi_priv *fsi, int is_play)
{
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
struct snd_soc_dai *dai = fsi_get_dai(io->substream);
+ struct fsi_master *master = fsi_get_master(fsi);
+ unsigned long flags;
+ spin_lock_irqsave(&master->lock, flags);
if (io->oerr_num > 0)
dev_err(dai->dev, "over_run = %d\n", io->oerr_num);
@@ -372,47 +424,27 @@
dev_err(dai->dev, "under_run = %d\n", io->uerr_num);
io->substream = NULL;
- io->buff_len = 0;
- io->buff_offset = 0;
- io->period_len = 0;
- io->period_num = 0;
+ io->buff_sample_capa = 0;
+ io->buff_sample_pos = 0;
+ io->period_samples = 0;
+ io->period_pos = 0;
io->oerr_num = 0;
io->uerr_num = 0;
+ spin_unlock_irqrestore(&master->lock, flags);
}
-static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play)
+static int fsi_get_current_fifo_samples(struct fsi_priv *fsi, int is_play)
{
u32 status;
- int data_num;
+ int frames;
status = is_play ?
fsi_reg_read(fsi, DOFF_ST) :
fsi_reg_read(fsi, DIFF_ST);
- data_num = 0x1ff & (status >> 8);
- data_num *= fsi->chan_num;
+ frames = 0x1ff & (status >> 8);
- return data_num;
-}
-
-static int fsi_len2num(int len, int width)
-{
- return len / width;
-}
-
-#define fsi_num2offset(a, b) fsi_num2len(a, b)
-static int fsi_num2len(int num, int width)
-{
- return num * width;
-}
-
-static int fsi_get_frame_width(struct fsi_priv *fsi, int is_play)
-{
- struct fsi_stream *io = fsi_get_stream(fsi, is_play);
- struct snd_pcm_substream *substream = io->substream;
- struct snd_pcm_runtime *runtime = substream->runtime;
-
- return frames_to_bytes(runtime, 1) / fsi->chan_num;
+ return fsi_frame2sample(fsi, frames);
}
static void fsi_count_fifo_err(struct fsi_priv *fsi)
@@ -444,8 +476,10 @@
{
int is_play = fsi_stream_is_play(stream);
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
+ struct snd_pcm_runtime *runtime = io->substream->runtime;
- return io->substream->runtime->dma_area + io->buff_offset;
+ return runtime->dma_area +
+ samples_to_bytes(runtime, io->buff_sample_pos);
}
static void fsi_dma_soft_push16(struct fsi_priv *fsi, int num)
@@ -559,37 +593,94 @@
/*
* clock function
*/
-#define fsi_module_init(m, d) __fsi_module_clk_ctrl(m, d, 1)
-#define fsi_module_kill(m, d) __fsi_module_clk_ctrl(m, d, 0)
-static void __fsi_module_clk_ctrl(struct fsi_master *master,
- struct device *dev,
- int enable)
-{
- pm_runtime_get_sync(dev);
-
- if (enable) {
- /* enable only SR */
- fsi_master_mask_set(master, SOFT_RST, FSISR, FSISR);
- fsi_master_mask_set(master, SOFT_RST, PASR | PBSR, 0);
- } else {
- /* clear all registers */
- fsi_master_mask_set(master, SOFT_RST, FSISR, 0);
- }
-
- pm_runtime_put_sync(dev);
-}
-
-#define fsi_port_start(f) __fsi_port_clk_ctrl(f, 1)
-#define fsi_port_stop(f) __fsi_port_clk_ctrl(f, 0)
-static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int enable)
+static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
+ long rate, int enable)
{
struct fsi_master *master = fsi_get_master(fsi);
- u32 soft = fsi_is_port_a(fsi) ? PASR : PBSR;
- u32 clk = fsi_is_port_a(fsi) ? CRA : CRB;
- int is_master = fsi_is_clk_master(fsi);
+ set_rate_func set_rate = fsi_get_info_set_rate(master);
+ int fsi_ver = master->core->ver;
+ int ret;
- fsi_master_mask_set(master, SOFT_RST, soft, (enable) ? soft : 0);
- if (is_master)
+ ret = set_rate(dev, fsi_is_port_a(fsi), rate, enable);
+ if (ret < 0) /* error */
+ return ret;
+
+ if (!enable)
+ return 0;
+
+ if (ret > 0) {
+ u32 data = 0;
+
+ switch (ret & SH_FSI_ACKMD_MASK) {
+ default:
+ /* FALL THROUGH */
+ case SH_FSI_ACKMD_512:
+ data |= (0x0 << 12);
+ break;
+ case SH_FSI_ACKMD_256:
+ data |= (0x1 << 12);
+ break;
+ case SH_FSI_ACKMD_128:
+ data |= (0x2 << 12);
+ break;
+ case SH_FSI_ACKMD_64:
+ data |= (0x3 << 12);
+ break;
+ case SH_FSI_ACKMD_32:
+ if (fsi_ver < 2)
+ dev_err(dev, "unsupported ACKMD\n");
+ else
+ data |= (0x4 << 12);
+ break;
+ }
+
+ switch (ret & SH_FSI_BPFMD_MASK) {
+ default:
+ /* FALL THROUGH */
+ case SH_FSI_BPFMD_32:
+ data |= (0x0 << 8);
+ break;
+ case SH_FSI_BPFMD_64:
+ data |= (0x1 << 8);
+ break;
+ case SH_FSI_BPFMD_128:
+ data |= (0x2 << 8);
+ break;
+ case SH_FSI_BPFMD_256:
+ data |= (0x3 << 8);
+ break;
+ case SH_FSI_BPFMD_512:
+ data |= (0x4 << 8);
+ break;
+ case SH_FSI_BPFMD_16:
+ if (fsi_ver < 2)
+ dev_err(dev, "unsupported ACKMD\n");
+ else
+ data |= (0x7 << 8);
+ break;
+ }
+
+ fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data);
+ udelay(10);
+ ret = 0;
+ }
+
+ return ret;
+}
+
+#define fsi_port_start(f, i) __fsi_port_clk_ctrl(f, i, 1)
+#define fsi_port_stop(f, i) __fsi_port_clk_ctrl(f, i, 0)
+static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int is_play, int enable)
+{
+ struct fsi_master *master = fsi_get_master(fsi);
+ u32 clk = fsi_is_port_a(fsi) ? CRA : CRB;
+
+ if (enable)
+ fsi_irq_enable(fsi, is_play);
+ else
+ fsi_irq_disable(fsi, is_play);
+
+ if (fsi_is_clk_master(fsi))
fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
}
@@ -598,18 +689,19 @@
*/
static void fsi_fifo_init(struct fsi_priv *fsi,
int is_play,
- struct snd_soc_dai *dai)
+ struct device *dev)
{
struct fsi_master *master = fsi_get_master(fsi);
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
u32 shift, i;
+ int frame_capa;
/* get on-chip RAM capacity */
shift = fsi_master_read(master, FIFO_SZ);
shift >>= fsi_get_port_shift(fsi, is_play);
shift &= FIFO_SZ_MASK;
- io->fifo_max_num = 256 << shift;
- dev_dbg(dai->dev, "fifo = %d words\n", io->fifo_max_num);
+ frame_capa = 256 << shift;
+ dev_dbg(dev, "fifo = %d words\n", frame_capa);
/*
* The maximum number of sample data varies depending
@@ -631,9 +723,11 @@
* 8 channels: 32 ( 32 x 8 = 256)
*/
for (i = 1; i < fsi->chan_num; i <<= 1)
- io->fifo_max_num >>= 1;
- dev_dbg(dai->dev, "%d channel %d store\n",
- fsi->chan_num, io->fifo_max_num);
+ frame_capa >>= 1;
+ dev_dbg(dev, "%d channel %d store\n",
+ fsi->chan_num, frame_capa);
+
+ io->fifo_sample_capa = fsi_frame2sample(fsi, frame_capa);
/*
* set interrupt generation factor
@@ -654,10 +748,10 @@
struct snd_pcm_substream *substream = NULL;
int is_play = fsi_stream_is_play(stream);
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
- int data_residue_num;
- int data_num;
- int data_num_max;
- int ch_width;
+ int sample_residues;
+ int sample_width;
+ int samples;
+ int samples_max;
int over_period;
void (*fn)(struct fsi_priv *fsi, int size);
@@ -673,36 +767,35 @@
/* FSI FIFO has limit.
* So, this driver can not send periods data at a time
*/
- if (io->buff_offset >=
- fsi_num2offset(io->period_num + 1, io->period_len)) {
+ if (io->buff_sample_pos >=
+ io->period_samples * (io->period_pos + 1)) {
over_period = 1;
- io->period_num = (io->period_num + 1) % runtime->periods;
+ io->period_pos = (io->period_pos + 1) % runtime->periods;
- if (0 == io->period_num)
- io->buff_offset = 0;
+ if (0 == io->period_pos)
+ io->buff_sample_pos = 0;
}
- /* get 1 channel data width */
- ch_width = fsi_get_frame_width(fsi, is_play);
+ /* get 1 sample data width */
+ sample_width = samples_to_bytes(runtime, 1);
- /* get residue data number of alsa */
- data_residue_num = fsi_len2num(io->buff_len - io->buff_offset,
- ch_width);
+ /* get number of residue samples */
+ sample_residues = io->buff_sample_capa - io->buff_sample_pos;
if (is_play) {
/*
* for play-back
*
- * data_num_max : number of FSI fifo free space
- * data_num : number of ALSA residue data
+ * samples_max : number of FSI fifo free samples space
+ * samples : number of ALSA residue samples
*/
- data_num_max = io->fifo_max_num * fsi->chan_num;
- data_num_max -= fsi_get_fifo_data_num(fsi, is_play);
+ samples_max = io->fifo_sample_capa;
+ samples_max -= fsi_get_current_fifo_samples(fsi, is_play);
- data_num = data_residue_num;
+ samples = sample_residues;
- switch (ch_width) {
+ switch (sample_width) {
case 2:
fn = fsi_dma_soft_push16;
break;
@@ -716,13 +809,13 @@
/*
* for capture
*
- * data_num_max : number of ALSA free space
- * data_num : number of data in FSI fifo
+ * samples_max : number of ALSA free samples space
+ * samples : number of samples in FSI fifo
*/
- data_num_max = data_residue_num;
- data_num = fsi_get_fifo_data_num(fsi, is_play);
+ samples_max = sample_residues;
+ samples = fsi_get_current_fifo_samples(fsi, is_play);
- switch (ch_width) {
+ switch (sample_width) {
case 2:
fn = fsi_dma_soft_pop16;
break;
@@ -734,12 +827,12 @@
}
}
- data_num = min(data_num, data_num_max);
+ samples = min(samples, samples_max);
- fn(fsi, data_num);
+ fn(fsi, samples);
- /* update buff_offset */
- io->buff_offset += fsi_num2offset(data_num, ch_width);
+ /* update buff_sample_pos */
+ io->buff_sample_pos += samples;
if (over_period)
snd_pcm_period_elapsed(substream);
@@ -788,16 +881,20 @@
* dai ops
*/
-static int fsi_dai_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
+static int fsi_hw_startup(struct fsi_priv *fsi,
+ int is_play,
+ struct device *dev)
{
- struct fsi_priv *fsi = fsi_get_priv(substream);
u32 flags = fsi_get_info_flags(fsi);
- u32 data;
- int is_play = fsi_is_play(substream);
+ u32 data = 0;
- pm_runtime_get_sync(dai->dev);
+ pm_runtime_get_sync(dev);
+ /* clock setting */
+ if (fsi_is_clk_master(fsi))
+ data = DIMD | DOMD;
+
+ fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data);
/* clock inversion (CKG2) */
data = 0;
@@ -812,54 +909,70 @@
fsi_reg_write(fsi, CKG2, data);
+ /* set format */
+ fsi_reg_write(fsi, DO_FMT, fsi->do_fmt);
+ fsi_reg_write(fsi, DI_FMT, fsi->di_fmt);
+
+ /* spdif ? */
+ if (fsi_is_spdif(fsi)) {
+ fsi_spdif_clk_ctrl(fsi, 1);
+ fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
+ }
+
/* irq clear */
fsi_irq_disable(fsi, is_play);
fsi_irq_clear_status(fsi);
/* fifo init */
- fsi_fifo_init(fsi, is_play, dai);
+ fsi_fifo_init(fsi, is_play, dev);
return 0;
}
+static void fsi_hw_shutdown(struct fsi_priv *fsi,
+ int is_play,
+ struct device *dev)
+{
+ if (fsi_is_clk_master(fsi))
+ fsi_set_master_clk(dev, fsi, fsi->rate, 0);
+
+ pm_runtime_put_sync(dev);
+}
+
+static int fsi_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct fsi_priv *fsi = fsi_get_priv(substream);
+ int is_play = fsi_is_play(substream);
+
+ return fsi_hw_startup(fsi, is_play, dai->dev);
+}
+
static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
int is_play = fsi_is_play(substream);
- struct fsi_master *master = fsi_get_master(fsi);
- set_rate_func set_rate = fsi_get_info_set_rate(master);
- fsi_irq_disable(fsi, is_play);
-
- if (fsi_is_clk_master(fsi))
- set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0);
-
+ fsi_hw_shutdown(fsi, is_play, dai->dev);
fsi->rate = 0;
-
- pm_runtime_put_sync(dai->dev);
}
static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
int is_play = fsi_is_play(substream);
int ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- fsi_stream_push(fsi, is_play, substream,
- frames_to_bytes(runtime, runtime->buffer_size),
- frames_to_bytes(runtime, runtime->period_size));
+ fsi_stream_push(fsi, is_play, substream);
ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi);
- fsi_irq_enable(fsi, is_play);
- fsi_port_start(fsi);
+ fsi_port_start(fsi, is_play);
break;
case SNDRV_PCM_TRIGGER_STOP:
- fsi_port_stop(fsi);
- fsi_irq_disable(fsi, is_play);
+ fsi_port_stop(fsi, is_play);
fsi_stream_pop(fsi, is_play);
break;
}
@@ -884,8 +997,8 @@
return -EINVAL;
}
- fsi_reg_write(fsi, DO_FMT, data);
- fsi_reg_write(fsi, DI_FMT, data);
+ fsi->do_fmt = data;
+ fsi->di_fmt = data;
return 0;
}
@@ -900,11 +1013,10 @@
data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM;
fsi->chan_num = 2;
- fsi_spdif_clk_ctrl(fsi, 1);
- fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
+ fsi->spdif = 1;
- fsi_reg_write(fsi, DO_FMT, data);
- fsi_reg_write(fsi, DI_FMT, data);
+ fsi->do_fmt = data;
+ fsi->di_fmt = data;
return 0;
}
@@ -915,32 +1027,24 @@
struct fsi_master *master = fsi_get_master(fsi);
set_rate_func set_rate = fsi_get_info_set_rate(master);
u32 flags = fsi_get_info_flags(fsi);
- u32 data = 0;
int ret;
- pm_runtime_get_sync(dai->dev);
-
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
- data = DIMD | DOMD;
fsi->clk_master = 1;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
- ret = -EINVAL;
- goto set_fmt_exit;
+ return -EINVAL;
}
if (fsi_is_clk_master(fsi) && !set_rate) {
dev_err(dai->dev, "platform doesn't have set_rate\n");
- ret = -EINVAL;
- goto set_fmt_exit;
+ return -EINVAL;
}
- fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data);
-
/* set format */
switch (flags & SH_FSI_FMT_MASK) {
case SH_FSI_FMT_DAI:
@@ -953,9 +1057,6 @@
ret = -EINVAL;
}
-set_fmt_exit:
- pm_runtime_put_sync(dai->dev);
-
return ret;
}
@@ -964,79 +1065,19 @@
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
- struct fsi_master *master = fsi_get_master(fsi);
- set_rate_func set_rate = fsi_get_info_set_rate(master);
- int fsi_ver = master->core->ver;
long rate = params_rate(params);
int ret;
if (!fsi_is_clk_master(fsi))
return 0;
- ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1);
- if (ret < 0) /* error */
+ ret = fsi_set_master_clk(dai->dev, fsi, rate, 1);
+ if (ret < 0)
return ret;
fsi->rate = rate;
- if (ret > 0) {
- u32 data = 0;
-
- switch (ret & SH_FSI_ACKMD_MASK) {
- default:
- /* FALL THROUGH */
- case SH_FSI_ACKMD_512:
- data |= (0x0 << 12);
- break;
- case SH_FSI_ACKMD_256:
- data |= (0x1 << 12);
- break;
- case SH_FSI_ACKMD_128:
- data |= (0x2 << 12);
- break;
- case SH_FSI_ACKMD_64:
- data |= (0x3 << 12);
- break;
- case SH_FSI_ACKMD_32:
- if (fsi_ver < 2)
- dev_err(dai->dev, "unsupported ACKMD\n");
- else
- data |= (0x4 << 12);
- break;
- }
-
- switch (ret & SH_FSI_BPFMD_MASK) {
- default:
- /* FALL THROUGH */
- case SH_FSI_BPFMD_32:
- data |= (0x0 << 8);
- break;
- case SH_FSI_BPFMD_64:
- data |= (0x1 << 8);
- break;
- case SH_FSI_BPFMD_128:
- data |= (0x2 << 8);
- break;
- case SH_FSI_BPFMD_256:
- data |= (0x3 << 8);
- break;
- case SH_FSI_BPFMD_512:
- data |= (0x4 << 8);
- break;
- case SH_FSI_BPFMD_16:
- if (fsi_ver < 2)
- dev_err(dai->dev, "unsupported ACKMD\n");
- else
- data |= (0x7 << 8);
- break;
- }
-
- fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data);
- udelay(10);
- ret = 0;
- }
return ret;
-
}
static struct snd_soc_dai_ops fsi_dai_ops = {
@@ -1097,16 +1138,14 @@
static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
struct fsi_priv *fsi = fsi_get_priv(substream);
struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream));
- long location;
+ int samples_pos = io->buff_sample_pos - 1;
- location = (io->buff_offset - 1);
- if (location < 0)
- location = 0;
+ if (samples_pos < 0)
+ samples_pos = 0;
- return bytes_to_frames(runtime, location);
+ return fsi_sample2frame(fsi, samples_pos);
}
static struct snd_pcm_ops fsi_pcm_ops = {
@@ -1129,10 +1168,10 @@
snd_pcm_lib_preallocate_free_for_all(pcm);
}
-static int fsi_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_pcm *pcm = rtd->pcm;
+
/*
* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
* in MMAP mode (i.e. aplay -M)
@@ -1246,8 +1285,6 @@
pm_runtime_enable(&pdev->dev);
dev_set_drvdata(&pdev->dev, master);
- fsi_module_init(master, &pdev->dev);
-
ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
id_entry->name, master);
if (ret) {
@@ -1290,8 +1327,6 @@
master = dev_get_drvdata(&pdev->dev);
- fsi_module_kill(master, &pdev->dev);
-
free_irq(master->irq, master);
pm_runtime_disable(&pdev->dev);
@@ -1305,53 +1340,43 @@
}
static void __fsi_suspend(struct fsi_priv *fsi,
- struct device *dev,
- set_rate_func set_rate)
+ int is_play,
+ struct device *dev)
{
- fsi->saved_do_fmt = fsi_reg_read(fsi, DO_FMT);
- fsi->saved_di_fmt = fsi_reg_read(fsi, DI_FMT);
- fsi->saved_ckg1 = fsi_reg_read(fsi, CKG1);
- fsi->saved_ckg2 = fsi_reg_read(fsi, CKG2);
- fsi->saved_out_sel = fsi_reg_read(fsi, OUT_SEL);
+ if (!fsi_stream_is_working(fsi, is_play))
+ return;
- if (fsi_is_clk_master(fsi))
- set_rate(dev, fsi_is_port_a(fsi), fsi->rate, 0);
+ fsi_port_stop(fsi, is_play);
+ fsi_hw_shutdown(fsi, is_play, dev);
}
static void __fsi_resume(struct fsi_priv *fsi,
- struct device *dev,
- set_rate_func set_rate)
+ int is_play,
+ struct device *dev)
{
- fsi_reg_write(fsi, DO_FMT, fsi->saved_do_fmt);
- fsi_reg_write(fsi, DI_FMT, fsi->saved_di_fmt);
- fsi_reg_write(fsi, CKG1, fsi->saved_ckg1);
- fsi_reg_write(fsi, CKG2, fsi->saved_ckg2);
- fsi_reg_write(fsi, OUT_SEL, fsi->saved_out_sel);
+ if (!fsi_stream_is_working(fsi, is_play))
+ return;
- if (fsi_is_clk_master(fsi))
- set_rate(dev, fsi_is_port_a(fsi), fsi->rate, 1);
+ fsi_hw_startup(fsi, is_play, dev);
+
+ if (fsi_is_clk_master(fsi) && fsi->rate)
+ fsi_set_master_clk(dev, fsi, fsi->rate, 1);
+
+ fsi_port_start(fsi, is_play);
+
}
static int fsi_suspend(struct device *dev)
{
struct fsi_master *master = dev_get_drvdata(dev);
- set_rate_func set_rate = fsi_get_info_set_rate(master);
+ struct fsi_priv *fsia = &master->fsia;
+ struct fsi_priv *fsib = &master->fsib;
- pm_runtime_get_sync(dev);
+ __fsi_suspend(fsia, 1, dev);
+ __fsi_suspend(fsia, 0, dev);
- __fsi_suspend(&master->fsia, dev, set_rate);
- __fsi_suspend(&master->fsib, dev, set_rate);
-
- master->saved_a_mclk = fsi_core_read(master, a_mclk);
- master->saved_b_mclk = fsi_core_read(master, b_mclk);
- master->saved_iemsk = fsi_core_read(master, iemsk);
- master->saved_imsk = fsi_core_read(master, imsk);
- master->saved_clk_rst = fsi_master_read(master, CLK_RST);
- master->saved_soft_rst = fsi_master_read(master, SOFT_RST);
-
- fsi_module_kill(master, dev);
-
- pm_runtime_put_sync(dev);
+ __fsi_suspend(fsib, 1, dev);
+ __fsi_suspend(fsib, 0, dev);
return 0;
}
@@ -1359,23 +1384,14 @@
static int fsi_resume(struct device *dev)
{
struct fsi_master *master = dev_get_drvdata(dev);
- set_rate_func set_rate = fsi_get_info_set_rate(master);
+ struct fsi_priv *fsia = &master->fsia;
+ struct fsi_priv *fsib = &master->fsib;
- pm_runtime_get_sync(dev);
+ __fsi_resume(fsia, 1, dev);
+ __fsi_resume(fsia, 0, dev);
- fsi_module_init(master, dev);
-
- fsi_master_mask_set(master, SOFT_RST, 0xffff, master->saved_soft_rst);
- fsi_master_mask_set(master, CLK_RST, 0xffff, master->saved_clk_rst);
- fsi_core_mask_set(master, a_mclk, 0xffff, master->saved_a_mclk);
- fsi_core_mask_set(master, b_mclk, 0xffff, master->saved_b_mclk);
- fsi_core_mask_set(master, iemsk, 0xffff, master->saved_iemsk);
- fsi_core_mask_set(master, imsk, 0xffff, master->saved_imsk);
-
- __fsi_resume(&master->fsia, dev, set_rate);
- __fsi_resume(&master->fsib, dev, set_rate);
-
- pm_runtime_put_sync(dev);
+ __fsi_resume(fsib, 1, dev);
+ __fsi_resume(fsib, 0, dev);
return 0;
}
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index a423bab..f8f6816 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -527,10 +527,11 @@
return bytes_to_frames(ss->runtime, ptr);
}
-static int siu_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int siu_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
/* card->dev == socdev->dev, see snd_soc_new_pcms() */
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
struct siu_info *info = siu_i2s_data;
struct platform_device *pdev = to_platform_device(card->dev);
int ret;
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 039b953..d9f8ade 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -20,422 +20,6 @@
#include <trace/events/asoc.h>
-#ifdef CONFIG_SPI_MASTER
-static int do_spi_write(void *control, const char *data, int len)
-{
- struct spi_device *spi = control;
- int ret;
-
- ret = spi_write(spi, data, len);
- if (ret < 0)
- return ret;
-
- return len;
-}
-#endif
-
-static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value, const void *data, int len)
-{
- int ret;
-
- if (!snd_soc_codec_volatile_register(codec, reg) &&
- reg < codec->driver->reg_cache_size &&
- !codec->cache_bypass) {
- ret = snd_soc_cache_write(codec, reg, value);
- if (ret < 0)
- return -1;
- }
-
- if (codec->cache_only) {
- codec->cache_sync = 1;
- return 0;
- }
-
- ret = codec->hw_write(codec->control_data, data, len);
- if (ret == len)
- return 0;
- if (ret < 0)
- return ret;
- else
- return -EIO;
-}
-
-static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg)
-{
- int ret;
- unsigned int val;
-
- if (reg >= codec->driver->reg_cache_size ||
- snd_soc_codec_volatile_register(codec, reg) ||
- codec->cache_bypass) {
- if (codec->cache_only)
- return -1;
-
- BUG_ON(!codec->hw_read);
- return codec->hw_read(codec, reg);
- }
-
- ret = snd_soc_cache_read(codec, reg, &val);
- if (ret < 0)
- return -1;
- return val;
-}
-
-static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u16 data;
-
- data = cpu_to_be16((reg << 12) | (value & 0xffffff));
-
- return do_hw_write(codec, reg, value, &data, 2);
-}
-
-static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- data[0] = (reg << 1) | ((value >> 8) & 0x0001);
- data[1] = value & 0x00ff;
-
- return do_hw_write(codec, reg, value, data, 2);
-}
-
-static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- reg &= 0xff;
- data[0] = reg;
- data[1] = value & 0xff;
-
- return do_hw_write(codec, reg, value, data, 2);
-}
-
-static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
-
- data[0] = reg;
- data[1] = (value >> 8) & 0xff;
- data[2] = value & 0xff;
-
- return do_hw_write(codec, reg, value, data, 3);
-}
-
-static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int do_i2c_read(struct snd_soc_codec *codec,
- void *reg, int reglen,
- void *data, int datalen)
-{
- struct i2c_msg xfer[2];
- int ret;
- struct i2c_client *client = codec->control_data;
-
- /* Write register */
- xfer[0].addr = client->addr;
- xfer[0].flags = 0;
- xfer[0].len = reglen;
- xfer[0].buf = reg;
-
- /* Read data */
- xfer[1].addr = client->addr;
- xfer[1].flags = I2C_M_RD;
- xfer[1].len = datalen;
- xfer[1].buf = data;
-
- ret = i2c_transfer(client->adapter, xfer, 2);
- if (ret == 2)
- return 0;
- else if (ret < 0)
- return ret;
- else
- return -EIO;
-}
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u8 reg = r;
- u8 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 1, &data, 1);
- if (ret < 0)
- return 0;
- return data;
-}
-#else
-#define snd_soc_8_8_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u8 reg = r;
- u16 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 1, &data, 2);
- if (ret < 0)
- return 0;
- return (data >> 8) | ((data & 0xff) << 8);
-}
-#else
-#define snd_soc_8_16_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u16 reg = r;
- u8 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 2, &data, 1);
- if (ret < 0)
- return 0;
- return data;
-}
-#else
-#define snd_soc_16_8_read_i2c NULL
-#endif
-
-static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
-
- data[0] = (reg >> 8) & 0xff;
- data[1] = reg & 0xff;
- data[2] = value;
-
- return do_hw_write(codec, reg, value, data, 3);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u16 reg = cpu_to_be16(r);
- u16 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 2, &data, 2);
- if (ret < 0)
- return 0;
- return be16_to_cpu(data);
-}
-#else
-#define snd_soc_16_16_read_i2c NULL
-#endif
-
-static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[4];
-
- data[0] = (reg >> 8) & 0xff;
- data[1] = reg & 0xff;
- data[2] = (value >> 8) & 0xff;
- data[3] = value & 0xff;
-
- return do_hw_write(codec, reg, value, data, 4);
-}
-
-/* Primitive bulk write support for soc-cache. The data pointed to by
- * `data' needs to already be in the form the hardware expects
- * including any leading register specific data. Any data written
- * through this function will not go through the cache as it only
- * handles writing to volatile or out of bounds registers.
- */
-static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg,
- const void *data, size_t len)
-{
- int ret;
-
- /* To ensure that we don't get out of sync with the cache, check
- * whether the base register is volatile or if we've directly asked
- * to bypass the cache. Out of bounds registers are considered
- * volatile.
- */
- if (!codec->cache_bypass
- && !snd_soc_codec_volatile_register(codec, reg)
- && reg < codec->driver->reg_cache_size)
- return -EINVAL;
-
- switch (codec->control_type) {
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
- case SND_SOC_I2C:
- ret = i2c_master_send(codec->control_data, data, len);
- break;
-#endif
-#if defined(CONFIG_SPI_MASTER)
- case SND_SOC_SPI:
- ret = spi_write(codec->control_data, data, len);
- break;
-#endif
- default:
- BUG();
- }
-
- if (ret == len)
- return 0;
- if (ret < 0)
- return ret;
- else
- return -EIO;
-}
-
-static struct {
- int addr_bits;
- int data_bits;
- int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
- unsigned int (*read)(struct snd_soc_codec *, unsigned int);
- unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
-} io_types[] = {
- {
- .addr_bits = 4, .data_bits = 12,
- .write = snd_soc_4_12_write, .read = snd_soc_4_12_read,
- },
- {
- .addr_bits = 7, .data_bits = 9,
- .write = snd_soc_7_9_write, .read = snd_soc_7_9_read,
- },
- {
- .addr_bits = 8, .data_bits = 8,
- .write = snd_soc_8_8_write, .read = snd_soc_8_8_read,
- .i2c_read = snd_soc_8_8_read_i2c,
- },
- {
- .addr_bits = 8, .data_bits = 16,
- .write = snd_soc_8_16_write, .read = snd_soc_8_16_read,
- .i2c_read = snd_soc_8_16_read_i2c,
- },
- {
- .addr_bits = 16, .data_bits = 8,
- .write = snd_soc_16_8_write, .read = snd_soc_16_8_read,
- .i2c_read = snd_soc_16_8_read_i2c,
- },
- {
- .addr_bits = 16, .data_bits = 16,
- .write = snd_soc_16_16_write, .read = snd_soc_16_16_read,
- .i2c_read = snd_soc_16_16_read_i2c,
- },
-};
-
-/**
- * snd_soc_codec_set_cache_io: Set up standard I/O functions.
- *
- * @codec: CODEC to configure.
- * @addr_bits: Number of bits of register address data.
- * @data_bits: Number of bits of data per register.
- * @control: Control bus used.
- *
- * Register formats are frequently shared between many I2C and SPI
- * devices. In order to promote code reuse the ASoC core provides
- * some standard implementations of CODEC read and write operations
- * which can be set up using this function.
- *
- * The caller is responsible for allocating and initialising the
- * actual cache.
- *
- * Note that at present this code cannot be used by CODECs with
- * volatile registers.
- */
-int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
- int addr_bits, int data_bits,
- enum snd_soc_control_type control)
-{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(io_types); i++)
- if (io_types[i].addr_bits == addr_bits &&
- io_types[i].data_bits == data_bits)
- break;
- if (i == ARRAY_SIZE(io_types)) {
- printk(KERN_ERR
- "No I/O functions for %d bit address %d bit data\n",
- addr_bits, data_bits);
- return -EINVAL;
- }
-
- codec->write = io_types[i].write;
- codec->read = io_types[i].read;
- codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
-
- switch (control) {
- case SND_SOC_I2C:
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
- codec->hw_write = (hw_write_t)i2c_master_send;
-#endif
- if (io_types[i].i2c_read)
- codec->hw_read = io_types[i].i2c_read;
-
- codec->control_data = container_of(codec->dev,
- struct i2c_client,
- dev);
- break;
-
- case SND_SOC_SPI:
-#ifdef CONFIG_SPI_MASTER
- codec->hw_write = do_spi_write;
-#endif
-
- codec->control_data = container_of(codec->dev,
- struct spi_device,
- dev);
- break;
- }
-
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
-
static bool snd_soc_set_cache_val(void *base, unsigned int idx,
unsigned int val, unsigned int word_size)
{
@@ -483,31 +67,86 @@
}
struct snd_soc_rbtree_node {
- struct rb_node node;
- unsigned int reg;
- unsigned int value;
- unsigned int defval;
+ struct rb_node node; /* the actual rbtree node holding this block */
+ unsigned int base_reg; /* base register handled by this block */
+ unsigned int word_size; /* number of bytes needed to represent the register index */
+ void *block; /* block of adjacent registers */
+ unsigned int blklen; /* number of registers available in the block */
} __attribute__ ((packed));
struct snd_soc_rbtree_ctx {
struct rb_root root;
+ struct snd_soc_rbtree_node *cached_rbnode;
};
+static inline void snd_soc_rbtree_get_base_top_reg(
+ struct snd_soc_rbtree_node *rbnode,
+ unsigned int *base, unsigned int *top)
+{
+ *base = rbnode->base_reg;
+ *top = rbnode->base_reg + rbnode->blklen - 1;
+}
+
+static unsigned int snd_soc_rbtree_get_register(
+ struct snd_soc_rbtree_node *rbnode, unsigned int idx)
+{
+ unsigned int val;
+
+ switch (rbnode->word_size) {
+ case 1: {
+ u8 *p = rbnode->block;
+ val = p[idx];
+ return val;
+ }
+ case 2: {
+ u16 *p = rbnode->block;
+ val = p[idx];
+ return val;
+ }
+ default:
+ BUG();
+ break;
+ }
+ return -1;
+}
+
+static void snd_soc_rbtree_set_register(struct snd_soc_rbtree_node *rbnode,
+ unsigned int idx, unsigned int val)
+{
+ switch (rbnode->word_size) {
+ case 1: {
+ u8 *p = rbnode->block;
+ p[idx] = val;
+ break;
+ }
+ case 2: {
+ u16 *p = rbnode->block;
+ p[idx] = val;
+ break;
+ }
+ default:
+ BUG();
+ break;
+ }
+}
+
static struct snd_soc_rbtree_node *snd_soc_rbtree_lookup(
struct rb_root *root, unsigned int reg)
{
struct rb_node *node;
struct snd_soc_rbtree_node *rbnode;
+ unsigned int base_reg, top_reg;
node = root->rb_node;
while (node) {
rbnode = container_of(node, struct snd_soc_rbtree_node, node);
- if (rbnode->reg < reg)
- node = node->rb_left;
- else if (rbnode->reg > reg)
- node = node->rb_right;
- else
+ snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
+ if (reg >= base_reg && reg <= top_reg)
return rbnode;
+ else if (reg > top_reg)
+ node = node->rb_right;
+ else if (reg < base_reg)
+ node = node->rb_left;
}
return NULL;
@@ -518,19 +157,28 @@
{
struct rb_node **new, *parent;
struct snd_soc_rbtree_node *rbnode_tmp;
+ unsigned int base_reg_tmp, top_reg_tmp;
+ unsigned int base_reg;
parent = NULL;
new = &root->rb_node;
while (*new) {
rbnode_tmp = container_of(*new, struct snd_soc_rbtree_node,
node);
+ /* base and top registers of the current rbnode */
+ snd_soc_rbtree_get_base_top_reg(rbnode_tmp, &base_reg_tmp,
+ &top_reg_tmp);
+ /* base register of the rbnode to be added */
+ base_reg = rbnode->base_reg;
parent = *new;
- if (rbnode_tmp->reg < rbnode->reg)
- new = &((*new)->rb_left);
- else if (rbnode_tmp->reg > rbnode->reg)
- new = &((*new)->rb_right);
- else
+ /* if this register has already been inserted, just return */
+ if (base_reg >= base_reg_tmp &&
+ base_reg <= top_reg_tmp)
return 0;
+ else if (base_reg > top_reg_tmp)
+ new = &((*new)->rb_right);
+ else if (base_reg < base_reg_tmp)
+ new = &((*new)->rb_left);
}
/* insert the node into the rbtree */
@@ -545,58 +193,146 @@
struct snd_soc_rbtree_ctx *rbtree_ctx;
struct rb_node *node;
struct snd_soc_rbtree_node *rbnode;
- unsigned int val;
+ unsigned int regtmp;
+ unsigned int val, def;
int ret;
+ int i;
rbtree_ctx = codec->reg_cache;
for (node = rb_first(&rbtree_ctx->root); node; node = rb_next(node)) {
rbnode = rb_entry(node, struct snd_soc_rbtree_node, node);
- if (rbnode->value == rbnode->defval)
- continue;
- WARN_ON(codec->writable_register &&
- codec->writable_register(codec, rbnode->reg));
- ret = snd_soc_cache_read(codec, rbnode->reg, &val);
- if (ret)
- return ret;
- codec->cache_bypass = 1;
- ret = snd_soc_write(codec, rbnode->reg, val);
- codec->cache_bypass = 0;
- if (ret)
- return ret;
- dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
- rbnode->reg, val);
+ for (i = 0; i < rbnode->blklen; ++i) {
+ regtmp = rbnode->base_reg + i;
+ WARN_ON(codec->writable_register &&
+ codec->writable_register(codec, regtmp));
+ val = snd_soc_rbtree_get_register(rbnode, i);
+ def = snd_soc_get_cache_val(codec->reg_def_copy, i,
+ rbnode->word_size);
+ if (val == def)
+ continue;
+
+ codec->cache_bypass = 1;
+ ret = snd_soc_write(codec, regtmp, val);
+ codec->cache_bypass = 0;
+ if (ret)
+ return ret;
+ dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
+ regtmp, val);
+ }
}
return 0;
}
+static int snd_soc_rbtree_insert_to_block(struct snd_soc_rbtree_node *rbnode,
+ unsigned int pos, unsigned int reg,
+ unsigned int value)
+{
+ u8 *blk;
+
+ blk = krealloc(rbnode->block,
+ (rbnode->blklen + 1) * rbnode->word_size, GFP_KERNEL);
+ if (!blk)
+ return -ENOMEM;
+
+ /* insert the register value in the correct place in the rbnode block */
+ memmove(blk + (pos + 1) * rbnode->word_size,
+ blk + pos * rbnode->word_size,
+ (rbnode->blklen - pos) * rbnode->word_size);
+
+ /* update the rbnode block, its size and the base register */
+ rbnode->block = blk;
+ rbnode->blklen++;
+ if (!pos)
+ rbnode->base_reg = reg;
+
+ snd_soc_rbtree_set_register(rbnode, pos, value);
+ return 0;
+}
+
static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
struct snd_soc_rbtree_ctx *rbtree_ctx;
- struct snd_soc_rbtree_node *rbnode;
+ struct snd_soc_rbtree_node *rbnode, *rbnode_tmp;
+ struct rb_node *node;
+ unsigned int val;
+ unsigned int reg_tmp;
+ unsigned int base_reg, top_reg;
+ unsigned int pos;
+ int i;
+ int ret;
rbtree_ctx = codec->reg_cache;
+ /* look up the required register in the cached rbnode */
+ rbnode = rbtree_ctx->cached_rbnode;
+ if (rbnode) {
+ snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
+ if (reg >= base_reg && reg <= top_reg) {
+ reg_tmp = reg - base_reg;
+ val = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+ if (val == value)
+ return 0;
+ snd_soc_rbtree_set_register(rbnode, reg_tmp, value);
+ return 0;
+ }
+ }
+ /* if we can't locate it in the cached rbnode we'll have
+ * to traverse the rbtree looking for it.
+ */
rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg);
if (rbnode) {
- if (rbnode->value == value)
+ reg_tmp = reg - rbnode->base_reg;
+ val = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+ if (val == value)
return 0;
- rbnode->value = value;
+ snd_soc_rbtree_set_register(rbnode, reg_tmp, value);
+ rbtree_ctx->cached_rbnode = rbnode;
} else {
/* bail out early, no need to create the rbnode yet */
if (!value)
return 0;
- /*
- * for uninitialized registers whose value is changed
- * from the default zero, create an rbnode and insert
- * it into the tree.
+ /* look for an adjacent register to the one we are about to add */
+ for (node = rb_first(&rbtree_ctx->root); node;
+ node = rb_next(node)) {
+ rbnode_tmp = rb_entry(node, struct snd_soc_rbtree_node, node);
+ for (i = 0; i < rbnode_tmp->blklen; ++i) {
+ reg_tmp = rbnode_tmp->base_reg + i;
+ if (abs(reg_tmp - reg) != 1)
+ continue;
+ /* decide where in the block to place our register */
+ if (reg_tmp + 1 == reg)
+ pos = i + 1;
+ else
+ pos = i;
+ ret = snd_soc_rbtree_insert_to_block(rbnode_tmp, pos,
+ reg, value);
+ if (ret)
+ return ret;
+ rbtree_ctx->cached_rbnode = rbnode_tmp;
+ return 0;
+ }
+ }
+ /* we did not manage to find a place to insert it in an existing
+ * block so create a new rbnode with a single register in its block.
+ * This block will get populated further if any other adjacent
+ * registers get modified in the future.
*/
rbnode = kzalloc(sizeof *rbnode, GFP_KERNEL);
if (!rbnode)
return -ENOMEM;
- rbnode->reg = reg;
- rbnode->value = value;
+ rbnode->blklen = 1;
+ rbnode->base_reg = reg;
+ rbnode->word_size = codec->driver->reg_word_size;
+ rbnode->block = kmalloc(rbnode->blklen * rbnode->word_size,
+ GFP_KERNEL);
+ if (!rbnode->block) {
+ kfree(rbnode);
+ return -ENOMEM;
+ }
+ snd_soc_rbtree_set_register(rbnode, 0, value);
snd_soc_rbtree_insert(&rbtree_ctx->root, rbnode);
+ rbtree_ctx->cached_rbnode = rbnode;
}
return 0;
@@ -607,11 +343,28 @@
{
struct snd_soc_rbtree_ctx *rbtree_ctx;
struct snd_soc_rbtree_node *rbnode;
+ unsigned int base_reg, top_reg;
+ unsigned int reg_tmp;
rbtree_ctx = codec->reg_cache;
+ /* look up the required register in the cached rbnode */
+ rbnode = rbtree_ctx->cached_rbnode;
+ if (rbnode) {
+ snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
+ if (reg >= base_reg && reg <= top_reg) {
+ reg_tmp = reg - base_reg;
+ *value = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+ return 0;
+ }
+ }
+ /* if we can't locate it in the cached rbnode we'll have
+ * to traverse the rbtree looking for it.
+ */
rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg);
if (rbnode) {
- *value = rbnode->value;
+ reg_tmp = reg - rbnode->base_reg;
+ *value = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+ rbtree_ctx->cached_rbnode = rbnode;
} else {
/* uninitialized registers default to 0 */
*value = 0;
@@ -637,6 +390,7 @@
rbtree_node = rb_entry(next, struct snd_soc_rbtree_node, node);
next = rb_next(&rbtree_node->node);
rb_erase(&rbtree_node->node, &rbtree_ctx->root);
+ kfree(rbtree_node->block);
kfree(rbtree_node);
}
@@ -649,10 +403,9 @@
static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec)
{
- struct snd_soc_rbtree_node *rbtree_node;
struct snd_soc_rbtree_ctx *rbtree_ctx;
- unsigned int val;
unsigned int word_size;
+ unsigned int val;
int i;
int ret;
@@ -662,32 +415,27 @@
rbtree_ctx = codec->reg_cache;
rbtree_ctx->root = RB_ROOT;
+ rbtree_ctx->cached_rbnode = NULL;
if (!codec->reg_def_copy)
return 0;
- /*
- * populate the rbtree with the initialized registers. All other
- * registers will be inserted when they are first modified.
- */
word_size = codec->driver->reg_word_size;
for (i = 0; i < codec->driver->reg_cache_size; ++i) {
- val = snd_soc_get_cache_val(codec->reg_def_copy, i, word_size);
+ val = snd_soc_get_cache_val(codec->reg_def_copy, i,
+ word_size);
if (!val)
continue;
- rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL);
- if (!rbtree_node) {
- ret = -ENOMEM;
- snd_soc_cache_exit(codec);
- break;
- }
- rbtree_node->reg = i;
- rbtree_node->value = val;
- rbtree_node->defval = val;
- snd_soc_rbtree_insert(&rbtree_ctx->root, rbtree_node);
+ ret = snd_soc_rbtree_cache_write(codec, i, val);
+ if (ret)
+ goto err;
}
return 0;
+
+err:
+ snd_soc_cache_exit(codec);
+ return ret;
}
#ifdef CONFIG_SND_SOC_CACHE_LZO
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d75043e..32bc503 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -44,7 +44,6 @@
#define NAME_SIZE 32
-static DEFINE_MUTEX(pcm_mutex);
static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
#ifdef CONFIG_DEBUG_FS
@@ -58,7 +57,7 @@
static LIST_HEAD(platform_list);
static LIST_HEAD(codec_list);
-static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
+int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
/*
* This is a timeout to do a DAPM powerdown after a stream is closed().
@@ -485,552 +484,6 @@
}
#endif
-static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- if (!codec_dai->driver->symmetric_rates &&
- !cpu_dai->driver->symmetric_rates &&
- !rtd->dai_link->symmetric_rates)
- return 0;
-
- /* This can happen if multiple streams are starting simultaneously -
- * the second can need to get its constraints before the first has
- * picked a rate. Complain and allow the application to carry on.
- */
- if (!rtd->rate) {
- dev_warn(&rtd->dev,
- "Not enforcing symmetric_rates due to race\n");
- return 0;
- }
-
- dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate);
-
- ret = snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- rtd->rate, rtd->rate);
- if (ret < 0) {
- dev_err(&rtd->dev,
- "Unable to apply rate symmetry constraint: %d\n", ret);
- return ret;
- }
-
- return 0;
-}
-
-/*
- * Called by ALSA when a PCM substream is opened, the runtime->hw record is
- * then initialized and any private data can be allocated. This also calls
- * startup for the cpu DAI, platform, machine and codec DAI.
- */
-static int soc_pcm_open(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
- struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver;
- int ret = 0;
-
- mutex_lock(&pcm_mutex);
-
- /* startup the audio subsystem */
- if (cpu_dai->driver->ops->startup) {
- ret = cpu_dai->driver->ops->startup(substream, cpu_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: can't open interface %s\n",
- cpu_dai->name);
- goto out;
- }
- }
-
- if (platform->driver->ops && platform->driver->ops->open) {
- ret = platform->driver->ops->open(substream);
- if (ret < 0) {
- printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
- goto platform_err;
- }
- }
-
- if (codec_dai->driver->ops->startup) {
- ret = codec_dai->driver->ops->startup(substream, codec_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: can't open codec %s\n",
- codec_dai->name);
- goto codec_dai_err;
- }
- }
-
- if (rtd->dai_link->ops && rtd->dai_link->ops->startup) {
- ret = rtd->dai_link->ops->startup(substream);
- if (ret < 0) {
- printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name);
- goto machine_err;
- }
- }
-
- /* Check that the codec and cpu DAIs are compatible */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- runtime->hw.rate_min =
- max(codec_dai_drv->playback.rate_min,
- cpu_dai_drv->playback.rate_min);
- runtime->hw.rate_max =
- min(codec_dai_drv->playback.rate_max,
- cpu_dai_drv->playback.rate_max);
- runtime->hw.channels_min =
- max(codec_dai_drv->playback.channels_min,
- cpu_dai_drv->playback.channels_min);
- runtime->hw.channels_max =
- min(codec_dai_drv->playback.channels_max,
- cpu_dai_drv->playback.channels_max);
- runtime->hw.formats =
- codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats;
- runtime->hw.rates =
- codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates;
- if (codec_dai_drv->playback.rates
- & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
- runtime->hw.rates |= cpu_dai_drv->playback.rates;
- if (cpu_dai_drv->playback.rates
- & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
- runtime->hw.rates |= codec_dai_drv->playback.rates;
- } else {
- runtime->hw.rate_min =
- max(codec_dai_drv->capture.rate_min,
- cpu_dai_drv->capture.rate_min);
- runtime->hw.rate_max =
- min(codec_dai_drv->capture.rate_max,
- cpu_dai_drv->capture.rate_max);
- runtime->hw.channels_min =
- max(codec_dai_drv->capture.channels_min,
- cpu_dai_drv->capture.channels_min);
- runtime->hw.channels_max =
- min(codec_dai_drv->capture.channels_max,
- cpu_dai_drv->capture.channels_max);
- runtime->hw.formats =
- codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats;
- runtime->hw.rates =
- codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates;
- if (codec_dai_drv->capture.rates
- & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
- runtime->hw.rates |= cpu_dai_drv->capture.rates;
- if (cpu_dai_drv->capture.rates
- & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
- runtime->hw.rates |= codec_dai_drv->capture.rates;
- }
-
- ret = -EINVAL;
- snd_pcm_limit_hw_rates(runtime);
- if (!runtime->hw.rates) {
- printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
- codec_dai->name, cpu_dai->name);
- goto config_err;
- }
- if (!runtime->hw.formats) {
- printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
- codec_dai->name, cpu_dai->name);
- goto config_err;
- }
- if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
- runtime->hw.channels_min > runtime->hw.channels_max) {
- printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
- codec_dai->name, cpu_dai->name);
- goto config_err;
- }
-
- /* Symmetry only applies if we've already got an active stream. */
- if (cpu_dai->active || codec_dai->active) {
- ret = soc_pcm_apply_symmetry(substream);
- if (ret != 0)
- goto config_err;
- }
-
- pr_debug("asoc: %s <-> %s info:\n",
- codec_dai->name, cpu_dai->name);
- pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
- pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
- runtime->hw.channels_max);
- pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
- runtime->hw.rate_max);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active++;
- codec_dai->playback_active++;
- } else {
- cpu_dai->capture_active++;
- codec_dai->capture_active++;
- }
- cpu_dai->active++;
- codec_dai->active++;
- rtd->codec->active++;
- mutex_unlock(&pcm_mutex);
- return 0;
-
-config_err:
- if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
- rtd->dai_link->ops->shutdown(substream);
-
-machine_err:
- if (codec_dai->driver->ops->shutdown)
- codec_dai->driver->ops->shutdown(substream, codec_dai);
-
-codec_dai_err:
- if (platform->driver->ops && platform->driver->ops->close)
- platform->driver->ops->close(substream);
-
-platform_err:
- if (cpu_dai->driver->ops->shutdown)
- cpu_dai->driver->ops->shutdown(substream, cpu_dai);
-out:
- mutex_unlock(&pcm_mutex);
- return ret;
-}
-
-/*
- * Power down the audio subsystem pmdown_time msecs after close is called.
- * This is to ensure there are no pops or clicks in between any music tracks
- * due to DAPM power cycling.
- */
-static void close_delayed_work(struct work_struct *work)
-{
- struct snd_soc_pcm_runtime *rtd =
- container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
-
- mutex_lock(&pcm_mutex);
-
- pr_debug("pop wq checking: %s status: %s waiting: %s\n",
- codec_dai->driver->playback.stream_name,
- codec_dai->playback_active ? "active" : "inactive",
- codec_dai->pop_wait ? "yes" : "no");
-
- /* are we waiting on this codec DAI stream */
- if (codec_dai->pop_wait == 1) {
- codec_dai->pop_wait = 0;
- snd_soc_dapm_stream_event(rtd,
- codec_dai->driver->playback.stream_name,
- SND_SOC_DAPM_STREAM_STOP);
- }
-
- mutex_unlock(&pcm_mutex);
-}
-
-/*
- * Called by ALSA when a PCM substream is closed. Private data can be
- * freed here. The cpu DAI, codec DAI, machine and platform are also
- * shutdown.
- */
-static int soc_codec_close(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&pcm_mutex);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active--;
- codec_dai->playback_active--;
- } else {
- cpu_dai->capture_active--;
- codec_dai->capture_active--;
- }
-
- cpu_dai->active--;
- codec_dai->active--;
- codec->active--;
-
- /* Muting the DAC suppresses artifacts caused during digital
- * shutdown, for example from stopping clocks.
- */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- snd_soc_dai_digital_mute(codec_dai, 1);
-
- if (cpu_dai->driver->ops->shutdown)
- cpu_dai->driver->ops->shutdown(substream, cpu_dai);
-
- if (codec_dai->driver->ops->shutdown)
- codec_dai->driver->ops->shutdown(substream, codec_dai);
-
- if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
- rtd->dai_link->ops->shutdown(substream);
-
- if (platform->driver->ops && platform->driver->ops->close)
- platform->driver->ops->close(substream);
- cpu_dai->runtime = NULL;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* start delayed pop wq here for playback streams */
- codec_dai->pop_wait = 1;
- schedule_delayed_work(&rtd->delayed_work,
- msecs_to_jiffies(rtd->pmdown_time));
- } else {
- /* capture streams can be powered down now */
- snd_soc_dapm_stream_event(rtd,
- codec_dai->driver->capture.stream_name,
- SND_SOC_DAPM_STREAM_STOP);
- }
-
- mutex_unlock(&pcm_mutex);
- return 0;
-}
-
-/*
- * Called by ALSA when the PCM substream is prepared, can set format, sample
- * rate, etc. This function is non atomic and can be called multiple times,
- * it can refer to the runtime info.
- */
-static int soc_pcm_prepare(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret = 0;
-
- mutex_lock(&pcm_mutex);
-
- if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) {
- ret = rtd->dai_link->ops->prepare(substream);
- if (ret < 0) {
- printk(KERN_ERR "asoc: machine prepare error\n");
- goto out;
- }
- }
-
- if (platform->driver->ops && platform->driver->ops->prepare) {
- ret = platform->driver->ops->prepare(substream);
- if (ret < 0) {
- printk(KERN_ERR "asoc: platform prepare error\n");
- goto out;
- }
- }
-
- if (codec_dai->driver->ops->prepare) {
- ret = codec_dai->driver->ops->prepare(substream, codec_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: codec DAI prepare error\n");
- goto out;
- }
- }
-
- if (cpu_dai->driver->ops->prepare) {
- ret = cpu_dai->driver->ops->prepare(substream, cpu_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: cpu DAI prepare error\n");
- goto out;
- }
- }
-
- /* cancel any delayed stream shutdown that is pending */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
- codec_dai->pop_wait) {
- codec_dai->pop_wait = 0;
- cancel_delayed_work(&rtd->delayed_work);
- }
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- snd_soc_dapm_stream_event(rtd,
- codec_dai->driver->playback.stream_name,
- SND_SOC_DAPM_STREAM_START);
- else
- snd_soc_dapm_stream_event(rtd,
- codec_dai->driver->capture.stream_name,
- SND_SOC_DAPM_STREAM_START);
-
- snd_soc_dai_digital_mute(codec_dai, 0);
-
-out:
- mutex_unlock(&pcm_mutex);
- return ret;
-}
-
-/*
- * Called by ALSA when the hardware params are set by application. This
- * function can also be called multiple times and can allocate buffers
- * (using snd_pcm_lib_* ). It's non-atomic.
- */
-static int soc_pcm_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_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret = 0;
-
- mutex_lock(&pcm_mutex);
-
- if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) {
- ret = rtd->dai_link->ops->hw_params(substream, params);
- if (ret < 0) {
- printk(KERN_ERR "asoc: machine hw_params failed\n");
- goto out;
- }
- }
-
- if (codec_dai->driver->ops->hw_params) {
- ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: can't set codec %s hw params\n",
- codec_dai->name);
- goto codec_err;
- }
- }
-
- if (cpu_dai->driver->ops->hw_params) {
- ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: interface %s hw params failed\n",
- cpu_dai->name);
- goto interface_err;
- }
- }
-
- if (platform->driver->ops && platform->driver->ops->hw_params) {
- ret = platform->driver->ops->hw_params(substream, params);
- if (ret < 0) {
- printk(KERN_ERR "asoc: platform %s hw params failed\n",
- platform->name);
- goto platform_err;
- }
- }
-
- rtd->rate = params_rate(params);
-
-out:
- mutex_unlock(&pcm_mutex);
- return ret;
-
-platform_err:
- if (cpu_dai->driver->ops->hw_free)
- cpu_dai->driver->ops->hw_free(substream, cpu_dai);
-
-interface_err:
- if (codec_dai->driver->ops->hw_free)
- codec_dai->driver->ops->hw_free(substream, codec_dai);
-
-codec_err:
- if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
- rtd->dai_link->ops->hw_free(substream);
-
- mutex_unlock(&pcm_mutex);
- return ret;
-}
-
-/*
- * Frees resources allocated by hw_params, can be called multiple times
- */
-static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&pcm_mutex);
-
- /* apply codec digital mute */
- if (!codec->active)
- snd_soc_dai_digital_mute(codec_dai, 1);
-
- /* free any machine hw params */
- if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
- rtd->dai_link->ops->hw_free(substream);
-
- /* free any DMA resources */
- if (platform->driver->ops && platform->driver->ops->hw_free)
- platform->driver->ops->hw_free(substream);
-
- /* now free hw params for the DAIs */
- if (codec_dai->driver->ops->hw_free)
- codec_dai->driver->ops->hw_free(substream, codec_dai);
-
- if (cpu_dai->driver->ops->hw_free)
- cpu_dai->driver->ops->hw_free(substream, cpu_dai);
-
- mutex_unlock(&pcm_mutex);
- return 0;
-}
-
-static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- if (codec_dai->driver->ops->trigger) {
- ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai);
- if (ret < 0)
- return ret;
- }
-
- if (platform->driver->ops && platform->driver->ops->trigger) {
- ret = platform->driver->ops->trigger(substream, cmd);
- if (ret < 0)
- return ret;
- }
-
- if (cpu_dai->driver->ops->trigger) {
- ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai);
- if (ret < 0)
- return ret;
- }
- return 0;
-}
-
-/*
- * soc level wrapper for pointer callback
- * If cpu_dai, codec_dai, platform driver has the delay callback, than
- * the runtime->delay will be updated accordingly.
- */
-static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_uframes_t offset = 0;
- snd_pcm_sframes_t delay = 0;
-
- if (platform->driver->ops && platform->driver->ops->pointer)
- offset = platform->driver->ops->pointer(substream);
-
- if (cpu_dai->driver->ops->delay)
- delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
-
- if (codec_dai->driver->ops->delay)
- delay += codec_dai->driver->ops->delay(substream, codec_dai);
-
- if (platform->driver->delay)
- delay += platform->driver->delay(substream, codec_dai);
-
- runtime->delay = delay;
-
- return offset;
-}
-
-/* ASoC PCM operations */
-static struct snd_pcm_ops soc_pcm_ops = {
- .open = soc_pcm_open,
- .close = soc_codec_close,
- .hw_params = soc_pcm_hw_params,
- .hw_free = soc_pcm_hw_free,
- .prepare = soc_pcm_prepare,
- .trigger = soc_pcm_trigger,
- .pointer = soc_pcm_pointer,
-};
-
#ifdef CONFIG_PM_SLEEP
/* powers down audio subsystem for suspend */
int snd_soc_suspend(struct device *dev)
@@ -1256,7 +709,7 @@
int snd_soc_resume(struct device *dev)
{
struct snd_soc_card *card = dev_get_drvdata(dev);
- int i;
+ int i, ac97_control = 0;
/* AC97 devices might have other drivers hanging off them so
* need to resume immediately. Other drivers don't have that
@@ -1265,14 +718,15 @@
*/
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
- if (cpu_dai->driver->ac97_control) {
- dev_dbg(dev, "Resuming AC97 immediately\n");
- soc_resume_deferred(&card->deferred_resume_work);
- } else {
- dev_dbg(dev, "Scheduling resume work\n");
- if (!schedule_work(&card->deferred_resume_work))
- dev_err(dev, "resume work item may be lost\n");
- }
+ ac97_control |= cpu_dai->driver->ac97_control;
+ }
+ if (ac97_control) {
+ dev_dbg(dev, "Resuming AC97 immediately\n");
+ soc_resume_deferred(&card->deferred_resume_work);
+ } else {
+ dev_dbg(dev, "Scheduling resume work\n");
+ if (!schedule_work(&card->deferred_resume_work))
+ dev_err(dev, "resume work item may be lost\n");
}
return 0;
@@ -1393,7 +847,7 @@
module_put(codec->dev->driver->owner);
}
-static void soc_remove_dai_link(struct snd_soc_card *card, int num)
+static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
{
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
struct snd_soc_codec *codec = rtd->codec;
@@ -1410,7 +864,8 @@
}
/* remove the CODEC DAI */
- if (codec_dai && codec_dai->probed) {
+ if (codec_dai && codec_dai->probed &&
+ codec_dai->driver->remove_order == order) {
if (codec_dai->driver->remove) {
err = codec_dai->driver->remove(codec_dai);
if (err < 0)
@@ -1421,7 +876,8 @@
}
/* remove the platform */
- if (platform && platform->probed) {
+ if (platform && platform->probed &&
+ platform->driver->remove_order == order) {
if (platform->driver->remove) {
err = platform->driver->remove(platform);
if (err < 0)
@@ -1433,11 +889,13 @@
}
/* remove the CODEC */
- if (codec && codec->probed)
+ if (codec && codec->probed &&
+ codec->driver->remove_order == order)
soc_remove_codec(codec);
/* remove the cpu_dai */
- if (cpu_dai && cpu_dai->probed) {
+ if (cpu_dai && cpu_dai->probed &&
+ cpu_dai->driver->remove_order == order) {
if (cpu_dai->driver->remove) {
err = cpu_dai->driver->remove(cpu_dai);
if (err < 0)
@@ -1451,11 +909,13 @@
static void soc_remove_dai_links(struct snd_soc_card *card)
{
- int i;
+ int dai, order;
- for (i = 0; i < card->num_rtd; i++)
- soc_remove_dai_link(card, i);
-
+ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+ order++) {
+ for (dai = 0; dai < card->num_rtd; dai++)
+ soc_remove_dai_link(card, dai, order);
+ }
card->num_rtd = 0;
}
@@ -1572,6 +1032,7 @@
rtd->dev.parent = card->dev;
rtd->dev.release = rtd_release;
rtd->dev.init_name = name;
+ mutex_init(&rtd->pcm_mutex);
ret = device_register(&rtd->dev);
if (ret < 0) {
dev_err(card->dev,
@@ -1596,7 +1057,7 @@
return 0;
}
-static int soc_probe_dai_link(struct snd_soc_card *card, int num)
+static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
{
struct snd_soc_dai_link *dai_link = &card->dai_link[num];
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
@@ -1605,7 +1066,8 @@
struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
int ret;
- dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num);
+ dev_dbg(card->dev, "probe %s dai link %d late %d\n",
+ card->name, num, order);
/* config components */
codec_dai->codec = codec;
@@ -1617,7 +1079,8 @@
rtd->pmdown_time = pmdown_time;
/* probe the cpu_dai */
- if (!cpu_dai->probed) {
+ if (!cpu_dai->probed &&
+ cpu_dai->driver->probe_order == order) {
if (!try_module_get(cpu_dai->dev->driver->owner))
return -ENODEV;
@@ -1636,14 +1099,16 @@
}
/* probe the CODEC */
- if (!codec->probed) {
+ if (!codec->probed &&
+ codec->driver->probe_order == order) {
ret = soc_probe_codec(card, codec);
if (ret < 0)
return ret;
}
/* probe the platform */
- if (!platform->probed) {
+ if (!platform->probed &&
+ platform->driver->probe_order == order) {
if (!try_module_get(platform->dev->driver->owner))
return -ENODEV;
@@ -1662,7 +1127,7 @@
}
/* probe the CODEC DAI */
- if (!codec_dai->probed) {
+ if (!codec_dai->probed && codec_dai->driver->probe_order == order) {
if (codec_dai->driver->probe) {
ret = codec_dai->driver->probe(codec_dai);
if (ret < 0) {
@@ -1677,8 +1142,9 @@
list_add(&codec_dai->card_list, &card->dai_dev_list);
}
- /* DAPM dai link stream work */
- INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+ /* complete DAI probe during last probe */
+ if (order != SND_SOC_COMP_ORDER_LAST)
+ return 0;
ret = soc_post_component_init(card, codec, num, 0);
if (ret)
@@ -1817,7 +1283,7 @@
struct snd_soc_codec *codec;
struct snd_soc_codec_conf *codec_conf;
enum snd_soc_compress_type compress_type;
- int ret, i;
+ int ret, i, order;
mutex_lock(&card->mutex);
@@ -1895,12 +1361,16 @@
goto card_probe_error;
}
- for (i = 0; i < card->num_links; i++) {
- ret = soc_probe_dai_link(card, i);
- if (ret < 0) {
- pr_err("asoc: failed to instantiate card %s: %d\n",
+ /* early DAI link probe */
+ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+ order++) {
+ for (i = 0; i < card->num_links; i++) {
+ ret = soc_probe_dai_link(card, i, order);
+ if (ret < 0) {
+ pr_err("asoc: failed to instantiate card %s: %d\n",
card->name, ret);
- goto probe_dai_err;
+ goto probe_dai_err;
+ }
}
}
@@ -2095,67 +1565,6 @@
.remove = soc_remove,
};
-/* create a new pcm */
-static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_pcm *pcm;
- char new_name[64];
- int ret = 0, playback = 0, capture = 0;
-
- /* check client and interface hw capabilities */
- snprintf(new_name, sizeof(new_name), "%s %s-%d",
- rtd->dai_link->stream_name, codec_dai->name, num);
-
- if (codec_dai->driver->playback.channels_min)
- playback = 1;
- if (codec_dai->driver->capture.channels_min)
- capture = 1;
-
- dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name);
- ret = snd_pcm_new(rtd->card->snd_card, new_name,
- num, playback, capture, &pcm);
- if (ret < 0) {
- printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
- return ret;
- }
-
- rtd->pcm = pcm;
- pcm->private_data = rtd;
- if (platform->driver->ops) {
- soc_pcm_ops.mmap = platform->driver->ops->mmap;
- soc_pcm_ops.pointer = platform->driver->ops->pointer;
- soc_pcm_ops.ioctl = platform->driver->ops->ioctl;
- soc_pcm_ops.copy = platform->driver->ops->copy;
- soc_pcm_ops.silence = platform->driver->ops->silence;
- soc_pcm_ops.ack = platform->driver->ops->ack;
- soc_pcm_ops.page = platform->driver->ops->page;
- }
-
- if (playback)
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
-
- if (capture)
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
-
- if (platform->driver->pcm_new) {
- ret = platform->driver->pcm_new(rtd->card->snd_card,
- codec_dai, pcm);
- if (ret < 0) {
- pr_err("asoc: platform pcm constructor failed\n");
- return ret;
- }
- }
-
- pcm->private_free = platform->driver->pcm_free;
- printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
- cpu_dai->name);
- return ret;
-}
-
/**
* snd_soc_codec_volatile_register: Report if a register is volatile.
*
@@ -2322,7 +1731,7 @@
return ret;
old = ret;
- new = (old & ~mask) | value;
+ new = (old & ~mask) | (value & mask);
change = old != new;
if (change) {
ret = snd_soc_write(codec, reg, new);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 32ab7fc..ceb2ba4 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -124,6 +124,43 @@
return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
}
+static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg)
+{
+ if (w->codec)
+ return snd_soc_read(w->codec, reg);
+ return 0;
+}
+
+static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val)
+{
+ if (w->codec)
+ return snd_soc_write(w->codec, reg, val);
+ return 0;
+}
+
+static int soc_widget_update_bits(struct snd_soc_dapm_widget *w,
+ unsigned short reg, unsigned int mask, unsigned int value)
+{
+ int change;
+ unsigned int old, new;
+ int ret;
+
+ ret = soc_widget_read(w, reg);
+ if (ret < 0)
+ return ret;
+
+ old = ret;
+ new = (old & ~mask) | (value & mask);
+ change = old != new;
+ if (change) {
+ ret = soc_widget_write(w, reg, new);
+ if (ret < 0)
+ return ret;
+ }
+
+ return change;
+}
+
/**
* snd_soc_dapm_set_bias_level - set the bias level for the system
* @dapm: DAPM context
@@ -139,39 +176,26 @@
struct snd_soc_card *card = dapm->card;
int ret = 0;
- switch (level) {
- case SND_SOC_BIAS_ON:
- dev_dbg(dapm->dev, "Setting full bias\n");
- break;
- case SND_SOC_BIAS_PREPARE:
- dev_dbg(dapm->dev, "Setting bias prepare\n");
- break;
- case SND_SOC_BIAS_STANDBY:
- dev_dbg(dapm->dev, "Setting standby bias\n");
- break;
- case SND_SOC_BIAS_OFF:
- dev_dbg(dapm->dev, "Setting bias off\n");
- break;
- default:
- dev_err(dapm->dev, "Setting invalid bias %d\n", level);
- return -EINVAL;
- }
-
trace_snd_soc_bias_level_start(card, level);
if (card && card->set_bias_level)
- ret = card->set_bias_level(card, level);
- if (ret == 0) {
- if (dapm->codec && dapm->codec->driver->set_bias_level)
- ret = dapm->codec->driver->set_bias_level(dapm->codec, level);
+ ret = card->set_bias_level(card, dapm, level);
+ if (ret != 0)
+ goto out;
+
+ if (dapm->codec) {
+ if (dapm->codec->driver->set_bias_level)
+ ret = dapm->codec->driver->set_bias_level(dapm->codec,
+ level);
else
dapm->bias_level = level;
}
- if (ret == 0) {
- if (card && card->set_bias_level_post)
- ret = card->set_bias_level_post(card, level);
- }
+ if (ret != 0)
+ goto out;
+ if (card && card->set_bias_level_post)
+ ret = card->set_bias_level_post(card, dapm, level);
+out:
trace_snd_soc_bias_level_done(card, level);
return ret;
@@ -194,7 +218,7 @@
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
- val = snd_soc_read(w->codec, reg);
+ val = soc_widget_read(w, reg);
val = (val >> shift) & mask;
if ((invert && !val) || (!invert && val))
@@ -209,8 +233,8 @@
int val, item, bitmask;
for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
- ;
- val = snd_soc_read(w->codec, e->reg);
+ ;
+ val = soc_widget_read(w, e->reg);
item = (val >> e->shift_l) & (bitmask - 1);
p->connect = 0;
@@ -240,7 +264,7 @@
w->kcontrol_news[i].private_value;
int val, item;
- val = snd_soc_read(w->codec, e->reg);
+ val = soc_widget_read(w, e->reg);
val = (val >> e->shift_l) & e->mask;
for (item = 0; item < e->max; item++) {
if (val == e->values[item])
@@ -606,6 +630,9 @@
}
list_for_each_entry(path, &widget->sinks, list_source) {
+ if (path->weak)
+ continue;
+
if (path->walked)
continue;
@@ -656,6 +683,9 @@
}
list_for_each_entry(path, &widget->sources, list_sink) {
+ if (path->weak)
+ continue;
+
if (path->walked)
continue;
@@ -681,7 +711,7 @@
else
val = w->off_val;
- snd_soc_update_bits(w->codec, -(w->reg + 1),
+ soc_widget_update_bits(w, -(w->reg + 1),
w->mask << w->shift, val << w->shift);
return 0;
@@ -737,6 +767,9 @@
/* Check if one of our outputs is connected */
list_for_each_entry(path, &w->sinks, list_source) {
+ if (path->weak)
+ continue;
+
if (path->connected &&
!path->connected(path->source, path->sink))
continue;
@@ -885,11 +918,17 @@
}
if (reg >= 0) {
+ /* Any widget will do, they should all be updating the
+ * same register.
+ */
+ w = list_first_entry(pending, struct snd_soc_dapm_widget,
+ power_list);
+
pop_dbg(dapm->dev, card->pop_time,
"pop test : Applying 0x%x/0x%x to %x in %dms\n",
value, mask, reg, card->pop_time);
pop_wait(card->pop_time);
- snd_soc_update_bits(dapm->codec, reg, mask, value);
+ soc_widget_update_bits(w, reg, mask, value);
}
list_for_each_entry(w, pending, power_list) {
@@ -1041,16 +1080,17 @@
struct snd_soc_dapm_context *d = data;
int ret;
- if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) {
+ /* If we're off and we're not supposed to be go into STANDBY */
+ if (d->bias_level == SND_SOC_BIAS_OFF &&
+ d->target_bias_level != SND_SOC_BIAS_OFF) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
if (ret != 0)
dev_err(d->dev,
"Failed to turn on bias: %d\n", ret);
}
- /* If we're changing to all on or all off then prepare */
- if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) ||
- (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) {
+ /* Prepare for a STADDBY->ON or ON->STANDBY transition */
+ if (d->bias_level != d->target_bias_level) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE);
if (ret != 0)
dev_err(d->dev,
@@ -1067,7 +1107,9 @@
int ret;
/* If we just powered the last thing off drop to standby bias */
- if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) {
+ if (d->bias_level == SND_SOC_BIAS_PREPARE &&
+ (d->target_bias_level == SND_SOC_BIAS_STANDBY ||
+ d->target_bias_level == SND_SOC_BIAS_OFF)) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
if (ret != 0)
dev_err(d->dev, "Failed to apply standby bias: %d\n",
@@ -1075,14 +1117,16 @@
}
/* If we're in standby and can support bias off then do that */
- if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) {
+ if (d->bias_level == SND_SOC_BIAS_STANDBY &&
+ d->target_bias_level == SND_SOC_BIAS_OFF) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF);
if (ret != 0)
dev_err(d->dev, "Failed to turn off bias: %d\n", ret);
}
/* If we just powered up then move to active bias */
- if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) {
+ if (d->bias_level == SND_SOC_BIAS_PREPARE &&
+ d->target_bias_level == SND_SOC_BIAS_ON) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON);
if (ret != 0)
dev_err(d->dev, "Failed to apply active bias: %d\n",
@@ -1107,13 +1151,19 @@
LIST_HEAD(up_list);
LIST_HEAD(down_list);
LIST_HEAD(async_domain);
+ enum snd_soc_bias_level bias;
int power;
trace_snd_soc_dapm_start(card);
- list_for_each_entry(d, &card->dapm_list, list)
- if (d->n_widgets || d->codec == NULL)
- d->dev_power = 0;
+ list_for_each_entry(d, &card->dapm_list, list) {
+ if (d->n_widgets || d->codec == NULL) {
+ if (d->idle_bias_off)
+ d->target_bias_level = SND_SOC_BIAS_OFF;
+ else
+ d->target_bias_level = SND_SOC_BIAS_STANDBY;
+ }
+ }
/* Check which widgets we need to power and store them in
* lists indicating if they should be powered up or down.
@@ -1135,8 +1185,27 @@
power = w->power_check(w);
else
power = 1;
- if (power)
- w->dapm->dev_power = 1;
+
+ if (power) {
+ d = w->dapm;
+
+ /* Supplies and micbiases only bring
+ * the context up to STANDBY as unless
+ * something else is active and
+ * passing audio they generally don't
+ * require full power.
+ */
+ switch (w->id) {
+ case snd_soc_dapm_supply:
+ case snd_soc_dapm_micbias:
+ if (d->target_bias_level < SND_SOC_BIAS_STANDBY)
+ d->target_bias_level = SND_SOC_BIAS_STANDBY;
+ break;
+ default:
+ d->target_bias_level = SND_SOC_BIAS_ON;
+ break;
+ }
+ }
if (w->power == power)
continue;
@@ -1160,24 +1229,19 @@
switch (event) {
case SND_SOC_DAPM_STREAM_START:
case SND_SOC_DAPM_STREAM_RESUME:
- dapm->dev_power = 1;
+ dapm->target_bias_level = SND_SOC_BIAS_ON;
break;
case SND_SOC_DAPM_STREAM_STOP:
- dapm->dev_power = !!dapm->codec->active;
+ if (dapm->codec->active)
+ dapm->target_bias_level = SND_SOC_BIAS_ON;
+ else
+ dapm->target_bias_level = SND_SOC_BIAS_STANDBY;
break;
case SND_SOC_DAPM_STREAM_SUSPEND:
- dapm->dev_power = 0;
+ dapm->target_bias_level = SND_SOC_BIAS_STANDBY;
break;
case SND_SOC_DAPM_STREAM_NOP:
- switch (dapm->bias_level) {
- case SND_SOC_BIAS_STANDBY:
- case SND_SOC_BIAS_OFF:
- dapm->dev_power = 0;
- break;
- default:
- dapm->dev_power = 1;
- break;
- }
+ dapm->target_bias_level = dapm->bias_level;
break;
default:
break;
@@ -1185,12 +1249,12 @@
}
/* Force all contexts in the card to the same bias state */
- power = 0;
+ bias = SND_SOC_BIAS_OFF;
list_for_each_entry(d, &card->dapm_list, list)
- if (d->dev_power)
- power = 1;
+ if (d->target_bias_level > bias)
+ bias = d->target_bias_level;
list_for_each_entry(d, &card->dapm_list, list)
- d->dev_power = power;
+ d->target_bias_level = bias;
/* Run all the bias changes in parallel */
@@ -1794,6 +1858,84 @@
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
+static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route)
+{
+ struct snd_soc_dapm_widget *source = dapm_find_widget(dapm,
+ route->source,
+ true);
+ struct snd_soc_dapm_widget *sink = dapm_find_widget(dapm,
+ route->sink,
+ true);
+ struct snd_soc_dapm_path *path;
+ int count = 0;
+
+ if (!source) {
+ dev_err(dapm->dev, "Unable to find source %s for weak route\n",
+ route->source);
+ return -ENODEV;
+ }
+
+ if (!sink) {
+ dev_err(dapm->dev, "Unable to find sink %s for weak route\n",
+ route->sink);
+ return -ENODEV;
+ }
+
+ if (route->control || route->connected)
+ dev_warn(dapm->dev, "Ignoring control for weak route %s->%s\n",
+ route->source, route->sink);
+
+ list_for_each_entry(path, &source->sinks, list_source) {
+ if (path->sink == sink) {
+ path->weak = 1;
+ count++;
+ }
+ }
+
+ if (count == 0)
+ dev_err(dapm->dev, "No path found for weak route %s->%s\n",
+ route->source, route->sink);
+ if (count > 1)
+ dev_warn(dapm->dev, "%d paths found for weak route %s->%s\n",
+ count, route->source, route->sink);
+
+ return 0;
+}
+
+/**
+ * snd_soc_dapm_weak_routes - Mark routes between DAPM widgets as weak
+ * @dapm: DAPM context
+ * @route: audio routes
+ * @num: number of routes
+ *
+ * Mark existing routes matching those specified in the passed array
+ * as being weak, meaning that they are ignored for the purpose of
+ * power decisions. The main intended use case is for sidetone paths
+ * which couple audio between other independent paths if they are both
+ * active in order to make the combination work better at the user
+ * level but which aren't intended to be "used".
+ *
+ * Note that CODEC drivers should not use this as sidetone type paths
+ * can frequently also be used as bypass paths.
+ */
+int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route, int num)
+{
+ int i, err;
+ int ret = 0;
+
+ for (i = 0; i < num; i++) {
+ err = snd_soc_dapm_weak_route(dapm, route);
+ if (err)
+ ret = err;
+ route++;
+ }
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes);
+
/**
* snd_soc_dapm_new_widgets - add new dapm widgets
* @dapm: DAPM context
@@ -1865,7 +2007,7 @@
/* Read the initial power state from the device */
if (w->reg >= 0) {
- val = snd_soc_read(w->codec, w->reg);
+ val = soc_widget_read(w, w->reg);
val &= 1 << w->shift;
if (w->invert)
val = !val;
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
new file mode 100644
index 0000000..cca490c
--- /dev/null
+++ b/sound/soc/soc-io.c
@@ -0,0 +1,396 @@
+/*
+ * soc-io.c -- ASoC register I/O helpers
+ *
+ * Copyright 2009-2011 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.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/i2c.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include <trace/events/asoc.h>
+
+#ifdef CONFIG_SPI_MASTER
+static int do_spi_write(void *control, const char *data, int len)
+{
+ struct spi_device *spi = control;
+ int ret;
+
+ ret = spi_write(spi, data, len);
+ if (ret < 0)
+ return ret;
+
+ return len;
+}
+#endif
+
+static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value, const void *data, int len)
+{
+ int ret;
+
+ if (!snd_soc_codec_volatile_register(codec, reg) &&
+ reg < codec->driver->reg_cache_size &&
+ !codec->cache_bypass) {
+ ret = snd_soc_cache_write(codec, reg, value);
+ if (ret < 0)
+ return -1;
+ }
+
+ if (codec->cache_only) {
+ codec->cache_sync = 1;
+ return 0;
+ }
+
+ ret = codec->hw_write(codec->control_data, data, len);
+ if (ret == len)
+ return 0;
+ if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+
+static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ int ret;
+ unsigned int val;
+
+ if (reg >= codec->driver->reg_cache_size ||
+ snd_soc_codec_volatile_register(codec, reg) ||
+ codec->cache_bypass) {
+ if (codec->cache_only)
+ return -1;
+
+ BUG_ON(!codec->hw_read);
+ return codec->hw_read(codec, reg);
+ }
+
+ ret = snd_soc_cache_read(codec, reg, &val);
+ if (ret < 0)
+ return -1;
+ return val;
+}
+
+static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 data;
+
+ data = cpu_to_be16((reg << 12) | (value & 0xffffff));
+
+ return do_hw_write(codec, reg, value, &data, 2);
+}
+
+static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 data;
+
+ data = cpu_to_be16((reg << 9) | (value & 0x1ff));
+
+ return do_hw_write(codec, reg, value, &data, 2);
+}
+
+static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ reg &= 0xff;
+ data[0] = reg;
+ data[1] = value & 0xff;
+
+ return do_hw_write(codec, reg, value, data, 2);
+}
+
+static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+ u16 val = cpu_to_be16(value);
+
+ data[0] = reg;
+ memcpy(&data[1], &val, sizeof(val));
+
+ return do_hw_write(codec, reg, value, data, 3);
+}
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int do_i2c_read(struct snd_soc_codec *codec,
+ void *reg, int reglen,
+ void *data, int datalen)
+{
+ struct i2c_msg xfer[2];
+ int ret;
+ struct i2c_client *client = codec->control_data;
+
+ /* Write register */
+ xfer[0].addr = client->addr;
+ xfer[0].flags = 0;
+ xfer[0].len = reglen;
+ xfer[0].buf = reg;
+
+ /* Read data */
+ xfer[1].addr = client->addr;
+ xfer[1].flags = I2C_M_RD;
+ xfer[1].len = datalen;
+ xfer[1].buf = data;
+
+ ret = i2c_transfer(client->adapter, xfer, 2);
+ if (ret == 2)
+ return 0;
+ else if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+#endif
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ u8 reg = r;
+ u8 data;
+ int ret;
+
+ ret = do_i2c_read(codec, ®, 1, &data, 1);
+ if (ret < 0)
+ return 0;
+ return data;
+}
+#else
+#define snd_soc_8_8_read_i2c NULL
+#endif
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ u8 reg = r;
+ u16 data;
+ int ret;
+
+ ret = do_i2c_read(codec, ®, 1, &data, 2);
+ if (ret < 0)
+ return 0;
+ return (data >> 8) | ((data & 0xff) << 8);
+}
+#else
+#define snd_soc_8_16_read_i2c NULL
+#endif
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ u16 reg = r;
+ u8 data;
+ int ret;
+
+ ret = do_i2c_read(codec, ®, 2, &data, 1);
+ if (ret < 0)
+ return 0;
+ return data;
+}
+#else
+#define snd_soc_16_8_read_i2c NULL
+#endif
+
+static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+ u16 rval = cpu_to_be16(reg);
+
+ memcpy(data, &rval, sizeof(rval));
+ data[2] = value;
+
+ return do_hw_write(codec, reg, value, data, 3);
+}
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ u16 reg = cpu_to_be16(r);
+ u16 data;
+ int ret;
+
+ ret = do_i2c_read(codec, ®, 2, &data, 2);
+ if (ret < 0)
+ return 0;
+ return be16_to_cpu(data);
+}
+#else
+#define snd_soc_16_16_read_i2c NULL
+#endif
+
+static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 data[2];
+
+ data[0] = cpu_to_be16(reg);
+ data[1] = cpu_to_be16(value);
+
+ return do_hw_write(codec, reg, value, data, sizeof(data));
+}
+
+/* Primitive bulk write support for soc-cache. The data pointed to by
+ * `data' needs to already be in the form the hardware expects
+ * including any leading register specific data. Any data written
+ * through this function will not go through the cache as it only
+ * handles writing to volatile or out of bounds registers.
+ */
+static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg,
+ const void *data, size_t len)
+{
+ int ret;
+
+ /* To ensure that we don't get out of sync with the cache, check
+ * whether the base register is volatile or if we've directly asked
+ * to bypass the cache. Out of bounds registers are considered
+ * volatile.
+ */
+ if (!codec->cache_bypass
+ && !snd_soc_codec_volatile_register(codec, reg)
+ && reg < codec->driver->reg_cache_size)
+ return -EINVAL;
+
+ switch (codec->control_type) {
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+ case SND_SOC_I2C:
+ ret = i2c_master_send(to_i2c_client(codec->dev), data, len);
+ break;
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ case SND_SOC_SPI:
+ ret = spi_write(to_spi_device(codec->dev), data, len);
+ break;
+#endif
+ default:
+ BUG();
+ }
+
+ if (ret == len)
+ return 0;
+ if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+
+static struct {
+ int addr_bits;
+ int data_bits;
+ int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
+ unsigned int (*read)(struct snd_soc_codec *, unsigned int);
+ unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+} io_types[] = {
+ {
+ .addr_bits = 4, .data_bits = 12,
+ .write = snd_soc_4_12_write,
+ },
+ {
+ .addr_bits = 7, .data_bits = 9,
+ .write = snd_soc_7_9_write,
+ },
+ {
+ .addr_bits = 8, .data_bits = 8,
+ .write = snd_soc_8_8_write,
+ .i2c_read = snd_soc_8_8_read_i2c,
+ },
+ {
+ .addr_bits = 8, .data_bits = 16,
+ .write = snd_soc_8_16_write,
+ .i2c_read = snd_soc_8_16_read_i2c,
+ },
+ {
+ .addr_bits = 16, .data_bits = 8,
+ .write = snd_soc_16_8_write,
+ .i2c_read = snd_soc_16_8_read_i2c,
+ },
+ {
+ .addr_bits = 16, .data_bits = 16,
+ .write = snd_soc_16_16_write,
+ .i2c_read = snd_soc_16_16_read_i2c,
+ },
+};
+
+/**
+ * snd_soc_codec_set_cache_io: Set up standard I/O functions.
+ *
+ * @codec: CODEC to configure.
+ * @addr_bits: Number of bits of register address data.
+ * @data_bits: Number of bits of data per register.
+ * @control: Control bus used.
+ *
+ * Register formats are frequently shared between many I2C and SPI
+ * devices. In order to promote code reuse the ASoC core provides
+ * some standard implementations of CODEC read and write operations
+ * which can be set up using this function.
+ *
+ * The caller is responsible for allocating and initialising the
+ * actual cache.
+ *
+ * Note that at present this code cannot be used by CODECs with
+ * volatile registers.
+ */
+int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
+ int addr_bits, int data_bits,
+ enum snd_soc_control_type control)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(io_types); i++)
+ if (io_types[i].addr_bits == addr_bits &&
+ io_types[i].data_bits == data_bits)
+ break;
+ if (i == ARRAY_SIZE(io_types)) {
+ printk(KERN_ERR
+ "No I/O functions for %d bit address %d bit data\n",
+ addr_bits, data_bits);
+ return -EINVAL;
+ }
+
+ codec->write = io_types[i].write;
+ codec->read = hw_read;
+ codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
+
+ switch (control) {
+ case SND_SOC_I2C:
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+ codec->hw_write = (hw_write_t)i2c_master_send;
+#endif
+ if (io_types[i].i2c_read)
+ codec->hw_read = io_types[i].i2c_read;
+
+ codec->control_data = container_of(codec->dev,
+ struct i2c_client,
+ dev);
+ break;
+
+ case SND_SOC_SPI:
+#ifdef CONFIG_SPI_MASTER
+ codec->hw_write = do_spi_write;
+#endif
+
+ codec->control_data = container_of(codec->dev,
+ struct spi_device,
+ dev);
+ break;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
+
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
new file mode 100644
index 0000000..b575939
--- /dev/null
+++ b/sound/soc/soc-pcm.c
@@ -0,0 +1,639 @@
+/*
+ * soc-pcm.c -- ALSA SoC PCM
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ * Copyright (C) 2010 Slimlogic Ltd.
+ * Copyright (C) 2010 Texas Instruments Inc.
+ *
+ * Authors: Liam Girdwood <lrg@ti.com>
+ * Mark Brown <broonie@opensource.wolfsonmicro.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/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/workqueue.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+static DEFINE_MUTEX(pcm_mutex);
+
+static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ if (!codec_dai->driver->symmetric_rates &&
+ !cpu_dai->driver->symmetric_rates &&
+ !rtd->dai_link->symmetric_rates)
+ return 0;
+
+ /* This can happen if multiple streams are starting simultaneously -
+ * the second can need to get its constraints before the first has
+ * picked a rate. Complain and allow the application to carry on.
+ */
+ if (!rtd->rate) {
+ dev_warn(&rtd->dev,
+ "Not enforcing symmetric_rates due to race\n");
+ return 0;
+ }
+
+ dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate);
+
+ ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ rtd->rate, rtd->rate);
+ if (ret < 0) {
+ dev_err(&rtd->dev,
+ "Unable to apply rate symmetry constraint: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+/*
+ * Called by ALSA when a PCM substream is opened, the runtime->hw record is
+ * then initialized and any private data can be allocated. This also calls
+ * startup for the cpu DAI, platform, machine and codec DAI.
+ */
+static int soc_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
+ struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver;
+ int ret = 0;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ /* startup the audio subsystem */
+ if (cpu_dai->driver->ops->startup) {
+ ret = cpu_dai->driver->ops->startup(substream, cpu_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open interface %s\n",
+ cpu_dai->name);
+ goto out;
+ }
+ }
+
+ if (platform->driver->ops && platform->driver->ops->open) {
+ ret = platform->driver->ops->open(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
+ goto platform_err;
+ }
+ }
+
+ if (codec_dai->driver->ops->startup) {
+ ret = codec_dai->driver->ops->startup(substream, codec_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open codec %s\n",
+ codec_dai->name);
+ goto codec_dai_err;
+ }
+ }
+
+ if (rtd->dai_link->ops && rtd->dai_link->ops->startup) {
+ ret = rtd->dai_link->ops->startup(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name);
+ goto machine_err;
+ }
+ }
+
+ /* Check that the codec and cpu DAIs are compatible */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw.rate_min =
+ max(codec_dai_drv->playback.rate_min,
+ cpu_dai_drv->playback.rate_min);
+ runtime->hw.rate_max =
+ min(codec_dai_drv->playback.rate_max,
+ cpu_dai_drv->playback.rate_max);
+ runtime->hw.channels_min =
+ max(codec_dai_drv->playback.channels_min,
+ cpu_dai_drv->playback.channels_min);
+ runtime->hw.channels_max =
+ min(codec_dai_drv->playback.channels_max,
+ cpu_dai_drv->playback.channels_max);
+ runtime->hw.formats =
+ codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats;
+ runtime->hw.rates =
+ codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates;
+ if (codec_dai_drv->playback.rates
+ & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+ runtime->hw.rates |= cpu_dai_drv->playback.rates;
+ if (cpu_dai_drv->playback.rates
+ & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+ runtime->hw.rates |= codec_dai_drv->playback.rates;
+ } else {
+ runtime->hw.rate_min =
+ max(codec_dai_drv->capture.rate_min,
+ cpu_dai_drv->capture.rate_min);
+ runtime->hw.rate_max =
+ min(codec_dai_drv->capture.rate_max,
+ cpu_dai_drv->capture.rate_max);
+ runtime->hw.channels_min =
+ max(codec_dai_drv->capture.channels_min,
+ cpu_dai_drv->capture.channels_min);
+ runtime->hw.channels_max =
+ min(codec_dai_drv->capture.channels_max,
+ cpu_dai_drv->capture.channels_max);
+ runtime->hw.formats =
+ codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats;
+ runtime->hw.rates =
+ codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates;
+ if (codec_dai_drv->capture.rates
+ & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+ runtime->hw.rates |= cpu_dai_drv->capture.rates;
+ if (cpu_dai_drv->capture.rates
+ & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+ runtime->hw.rates |= codec_dai_drv->capture.rates;
+ }
+
+ ret = -EINVAL;
+ snd_pcm_limit_hw_rates(runtime);
+ if (!runtime->hw.rates) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
+ codec_dai->name, cpu_dai->name);
+ goto config_err;
+ }
+ if (!runtime->hw.formats) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
+ codec_dai->name, cpu_dai->name);
+ goto config_err;
+ }
+ if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
+ runtime->hw.channels_min > runtime->hw.channels_max) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
+ codec_dai->name, cpu_dai->name);
+ goto config_err;
+ }
+
+ /* Symmetry only applies if we've already got an active stream. */
+ if (cpu_dai->active || codec_dai->active) {
+ ret = soc_pcm_apply_symmetry(substream);
+ if (ret != 0)
+ goto config_err;
+ }
+
+ pr_debug("asoc: %s <-> %s info:\n",
+ codec_dai->name, cpu_dai->name);
+ pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
+ pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
+ runtime->hw.channels_max);
+ pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
+ runtime->hw.rate_max);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ cpu_dai->playback_active++;
+ codec_dai->playback_active++;
+ } else {
+ cpu_dai->capture_active++;
+ codec_dai->capture_active++;
+ }
+ cpu_dai->active++;
+ codec_dai->active++;
+ rtd->codec->active++;
+ mutex_unlock(&rtd->pcm_mutex);
+ return 0;
+
+config_err:
+ if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
+ rtd->dai_link->ops->shutdown(substream);
+
+machine_err:
+ if (codec_dai->driver->ops->shutdown)
+ codec_dai->driver->ops->shutdown(substream, codec_dai);
+
+codec_dai_err:
+ if (platform->driver->ops && platform->driver->ops->close)
+ platform->driver->ops->close(substream);
+
+platform_err:
+ if (cpu_dai->driver->ops->shutdown)
+ cpu_dai->driver->ops->shutdown(substream, cpu_dai);
+out:
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+}
+
+/*
+ * Power down the audio subsystem pmdown_time msecs after close is called.
+ * This is to ensure there are no pops or clicks in between any music tracks
+ * due to DAPM power cycling.
+ */
+static void close_delayed_work(struct work_struct *work)
+{
+ struct snd_soc_pcm_runtime *rtd =
+ container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ pr_debug("pop wq checking: %s status: %s waiting: %s\n",
+ codec_dai->driver->playback.stream_name,
+ codec_dai->playback_active ? "active" : "inactive",
+ codec_dai->pop_wait ? "yes" : "no");
+
+ /* are we waiting on this codec DAI stream */
+ if (codec_dai->pop_wait == 1) {
+ codec_dai->pop_wait = 0;
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
+ }
+
+ mutex_unlock(&rtd->pcm_mutex);
+}
+
+/*
+ * Called by ALSA when a PCM substream is closed. Private data can be
+ * freed here. The cpu DAI, codec DAI, machine and platform are also
+ * shutdown.
+ */
+static int soc_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_codec *codec = rtd->codec;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ cpu_dai->playback_active--;
+ codec_dai->playback_active--;
+ } else {
+ cpu_dai->capture_active--;
+ codec_dai->capture_active--;
+ }
+
+ cpu_dai->active--;
+ codec_dai->active--;
+ codec->active--;
+
+ /* Muting the DAC suppresses artifacts caused during digital
+ * shutdown, for example from stopping clocks.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dai_digital_mute(codec_dai, 1);
+
+ if (cpu_dai->driver->ops->shutdown)
+ cpu_dai->driver->ops->shutdown(substream, cpu_dai);
+
+ if (codec_dai->driver->ops->shutdown)
+ codec_dai->driver->ops->shutdown(substream, codec_dai);
+
+ if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
+ rtd->dai_link->ops->shutdown(substream);
+
+ if (platform->driver->ops && platform->driver->ops->close)
+ platform->driver->ops->close(substream);
+ cpu_dai->runtime = NULL;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* start delayed pop wq here for playback streams */
+ codec_dai->pop_wait = 1;
+ schedule_delayed_work(&rtd->delayed_work,
+ msecs_to_jiffies(rtd->pmdown_time));
+ } else {
+ /* capture streams can be powered down now */
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
+ }
+
+ mutex_unlock(&rtd->pcm_mutex);
+ return 0;
+}
+
+/*
+ * Called by ALSA when the PCM substream is prepared, can set format, sample
+ * rate, etc. This function is non atomic and can be called multiple times,
+ * it can refer to the runtime info.
+ */
+static int soc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret = 0;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) {
+ ret = rtd->dai_link->ops->prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: machine prepare error\n");
+ goto out;
+ }
+ }
+
+ if (platform->driver->ops && platform->driver->ops->prepare) {
+ ret = platform->driver->ops->prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: platform prepare error\n");
+ goto out;
+ }
+ }
+
+ if (codec_dai->driver->ops->prepare) {
+ ret = codec_dai->driver->ops->prepare(substream, codec_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: codec DAI prepare error\n");
+ goto out;
+ }
+ }
+
+ if (cpu_dai->driver->ops->prepare) {
+ ret = cpu_dai->driver->ops->prepare(substream, cpu_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: cpu DAI prepare error\n");
+ goto out;
+ }
+ }
+
+ /* cancel any delayed stream shutdown that is pending */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ codec_dai->pop_wait) {
+ codec_dai->pop_wait = 0;
+ cancel_delayed_work(&rtd->delayed_work);
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+
+ snd_soc_dai_digital_mute(codec_dai, 0);
+
+out:
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+}
+
+/*
+ * Called by ALSA when the hardware params are set by application. This
+ * function can also be called multiple times and can allocate buffers
+ * (using snd_pcm_lib_* ). It's non-atomic.
+ */
+static int soc_pcm_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_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret = 0;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) {
+ ret = rtd->dai_link->ops->hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: machine hw_params failed\n");
+ goto out;
+ }
+ }
+
+ if (codec_dai->driver->ops->hw_params) {
+ ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't set codec %s hw params\n",
+ codec_dai->name);
+ goto codec_err;
+ }
+ }
+
+ if (cpu_dai->driver->ops->hw_params) {
+ ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: interface %s hw params failed\n",
+ cpu_dai->name);
+ goto interface_err;
+ }
+ }
+
+ if (platform->driver->ops && platform->driver->ops->hw_params) {
+ ret = platform->driver->ops->hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: platform %s hw params failed\n",
+ platform->name);
+ goto platform_err;
+ }
+ }
+
+ rtd->rate = params_rate(params);
+
+out:
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+
+platform_err:
+ if (cpu_dai->driver->ops->hw_free)
+ cpu_dai->driver->ops->hw_free(substream, cpu_dai);
+
+interface_err:
+ if (codec_dai->driver->ops->hw_free)
+ codec_dai->driver->ops->hw_free(substream, codec_dai);
+
+codec_err:
+ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
+ rtd->dai_link->ops->hw_free(substream);
+
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+}
+
+/*
+ * Frees resources allocated by hw_params, can be called multiple times
+ */
+static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_codec *codec = rtd->codec;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ /* apply codec digital mute */
+ if (!codec->active)
+ snd_soc_dai_digital_mute(codec_dai, 1);
+
+ /* free any machine hw params */
+ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
+ rtd->dai_link->ops->hw_free(substream);
+
+ /* free any DMA resources */
+ if (platform->driver->ops && platform->driver->ops->hw_free)
+ platform->driver->ops->hw_free(substream);
+
+ /* now free hw params for the DAIs */
+ if (codec_dai->driver->ops->hw_free)
+ codec_dai->driver->ops->hw_free(substream, codec_dai);
+
+ if (cpu_dai->driver->ops->hw_free)
+ cpu_dai->driver->ops->hw_free(substream, cpu_dai);
+
+ mutex_unlock(&rtd->pcm_mutex);
+ return 0;
+}
+
+static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ if (codec_dai->driver->ops->trigger) {
+ ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (platform->driver->ops && platform->driver->ops->trigger) {
+ ret = platform->driver->ops->trigger(substream, cmd);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (cpu_dai->driver->ops->trigger) {
+ ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai);
+ if (ret < 0)
+ return ret;
+ }
+ return 0;
+}
+
+/*
+ * soc level wrapper for pointer callback
+ * If cpu_dai, codec_dai, platform driver has the delay callback, than
+ * the runtime->delay will be updated accordingly.
+ */
+static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t offset = 0;
+ snd_pcm_sframes_t delay = 0;
+
+ if (platform->driver->ops && platform->driver->ops->pointer)
+ offset = platform->driver->ops->pointer(substream);
+
+ if (cpu_dai->driver->ops->delay)
+ delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
+
+ if (codec_dai->driver->ops->delay)
+ delay += codec_dai->driver->ops->delay(substream, codec_dai);
+
+ if (platform->driver->delay)
+ delay += platform->driver->delay(substream, codec_dai);
+
+ runtime->delay = delay;
+
+ return offset;
+}
+
+/* ASoC PCM operations */
+static struct snd_pcm_ops soc_pcm_ops = {
+ .open = soc_pcm_open,
+ .close = soc_pcm_close,
+ .hw_params = soc_pcm_hw_params,
+ .hw_free = soc_pcm_hw_free,
+ .prepare = soc_pcm_prepare,
+ .trigger = soc_pcm_trigger,
+ .pointer = soc_pcm_pointer,
+};
+
+/* create a new pcm */
+int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_pcm *pcm;
+ char new_name[64];
+ int ret = 0, playback = 0, capture = 0;
+
+ /* check client and interface hw capabilities */
+ snprintf(new_name, sizeof(new_name), "%s %s-%d",
+ rtd->dai_link->stream_name, codec_dai->name, num);
+
+ if (codec_dai->driver->playback.channels_min)
+ playback = 1;
+ if (codec_dai->driver->capture.channels_min)
+ capture = 1;
+
+ dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name);
+ ret = snd_pcm_new(rtd->card->snd_card, new_name,
+ num, playback, capture, &pcm);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
+ return ret;
+ }
+
+ /* DAPM dai link stream work */
+ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+
+ rtd->pcm = pcm;
+ pcm->private_data = rtd;
+ if (platform->driver->ops) {
+ soc_pcm_ops.mmap = platform->driver->ops->mmap;
+ soc_pcm_ops.pointer = platform->driver->ops->pointer;
+ soc_pcm_ops.ioctl = platform->driver->ops->ioctl;
+ soc_pcm_ops.copy = platform->driver->ops->copy;
+ soc_pcm_ops.silence = platform->driver->ops->silence;
+ soc_pcm_ops.ack = platform->driver->ops->ack;
+ soc_pcm_ops.page = platform->driver->ops->page;
+ }
+
+ if (playback)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
+
+ if (capture)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
+
+ if (platform->driver->pcm_new) {
+ ret = platform->driver->pcm_new(rtd);
+ if (ret < 0) {
+ pr_err("asoc: platform pcm constructor failed\n");
+ return ret;
+ }
+ }
+
+ pcm->private_free = platform->driver->pcm_free;
+ printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
+ cpu_dai->name);
+ return ret;
+}
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 3c271f9..ff86e5e 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -322,9 +322,11 @@
static u64 tegra_dma_mask = DMA_BIT_MASK(32);
-static int tegra_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0d6738a..a42e9ac 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -267,7 +267,7 @@
}
machine->gpio_requested |= GPIO_HP_MUTE;
- gpio_direction_output(pdata->gpio_hp_mute, 0);
+ gpio_direction_output(pdata->gpio_hp_mute, 1);
}
if (gpio_is_valid(pdata->gpio_int_mic_en)) {
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index f4aa4e0..34aa972 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -288,9 +288,10 @@
snd_pcm_lib_preallocate_free_for_all(pcm);
}
-static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int txx9aclc_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
struct platform_device *pdev = to_platform_device(dai->platform->dev);
struct txx9aclc_soc_device *dev;
struct resource *r;