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-rw-r--r--Documentation/sound/alsa/soc/codec.txt45
-rw-r--r--Documentation/sound/alsa/soc/machine.txt38
-rw-r--r--Documentation/sound/alsa/soc/platform.txt12
-rw-r--r--sound/soc/codecs/Kconfig3
-rw-r--r--sound/soc/codecs/Makefile2
-rw-r--r--sound/soc/codecs/cs4270.c161
-rw-r--r--sound/soc/codecs/dmic.c81
-rw-r--r--sound/soc/codecs/tlv320dac33.c269
-rw-r--r--sound/soc/codecs/tpa6130a2.c1
-rw-r--r--sound/soc/codecs/twl6040.c8
-rw-r--r--sound/soc/codecs/wm8995.c2
-rw-r--r--sound/soc/omap/omap-mcbsp.c35
-rw-r--r--sound/soc/omap/omap-mcbsp.h4
-rw-r--r--sound/soc/samsung/rx1950_uda1380.c1
-rw-r--r--sound/soc/sh/migor.c2
-rw-r--r--sound/soc/soc-cache.c2
16 files changed, 326 insertions, 340 deletions
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
index 37ba3a72cb76..bce23a4a7875 100644
--- a/Documentation/sound/alsa/soc/codec.txt
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -27,42 +27,38 @@ ASoC Codec driver breakdown
1 - Codec DAI and PCM configuration
-----------------------------------
-Each codec driver must have a struct snd_soc_codec_dai to define its DAI and
+Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
PCM capabilities and operations. This struct is exported so that it can be
registered with the core by your machine driver.
e.g.
-struct snd_soc_codec_dai wm8731_dai = {
- .name = "WM8731",
- /* playback capabilities */
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+ .prepare = wm8731_pcm_prepare,
+ .hw_params = wm8731_hw_params,
+ .shutdown = wm8731_shutdown,
+ .digital_mute = wm8731_mute,
+ .set_sysclk = wm8731_set_dai_sysclk,
+ .set_fmt = wm8731_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver wm8731_dai = {
+ .name = "wm8731-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8731_RATES,
.formats = WM8731_FORMATS,},
- /* capture capabilities */
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8731_RATES,
.formats = WM8731_FORMATS,},
- /* pcm operations - see section 4 below */
- .ops = {
- .prepare = wm8731_pcm_prepare,
- .hw_params = wm8731_hw_params,
- .shutdown = wm8731_shutdown,
- },
- /* DAI operations - see DAI.txt */
- .dai_ops = {
- .digital_mute = wm8731_mute,
- .set_sysclk = wm8731_set_dai_sysclk,
- .set_fmt = wm8731_set_dai_fmt,
- }
+ .ops = &wm8731_dai_ops,
+ .symmetric_rates = 1,
};
-EXPORT_SYMBOL_GPL(wm8731_dai);
2 - Codec control IO
@@ -186,13 +182,14 @@ when the mute is applied or freed.
i.e.
-static int wm8974_mute(struct snd_soc_codec *codec,
- struct snd_soc_codec_dai *dai, int mute)
+static int wm8974_mute(struct snd_soc_dai *dai, int mute)
{
- u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf;
- if(mute)
- wm8974_write(codec, WM8974_DAC, mute_reg | 0x40);
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+ if (mute)
+ snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
else
- wm8974_write(codec, WM8974_DAC, mute_reg);
+ snd_soc_write(codec, WM8974_DAC, mute_reg);
return 0;
}
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index 2524c75557df..3e2ec9cbf397 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -12,6 +12,8 @@ the following struct:-
struct snd_soc_card {
char *name;
+ ...
+
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
@@ -22,12 +24,13 @@ struct snd_soc_card {
int (*resume_pre)(struct platform_device *pdev);
int (*resume_post)(struct platform_device *pdev);
- /* machine stream operations */
- struct snd_soc_ops *ops;
+ ...
/* CPU <--> Codec DAI links */
struct snd_soc_dai_link *dai_link;
int num_links;
+
+ ...
};
probe()/remove()
@@ -42,11 +45,6 @@ of any machine audio tasks that have to be done before or after the codec, DAIs
and DMA is suspended and resumed. Optional.
-Machine operations
-------------------
-The machine specific audio operations can be set here. Again this is optional.
-
-
Machine DAI Configuration
-------------------------
The machine DAI configuration glues all the codec and CPU DAIs together. It can
@@ -61,8 +59,10 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
static struct snd_soc_dai_link corgi_dai = {
.name = "WM8731",
.stream_name = "WM8731",
- .cpu_dai = &pxa_i2s_dai,
- .codec_dai = &wm8731_dai,
+ .cpu_dai_name = "pxa-is2-dai",
+ .codec_dai_name = "wm8731-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm8713-codec.0-001a",
.init = corgi_wm8731_init,
.ops = &corgi_ops,
};
@@ -77,26 +77,6 @@ static struct snd_soc_card snd_soc_corgi = {
};
-Machine Audio Subsystem
------------------------
-
-The machine soc device glues the platform, machine and codec driver together.
-Private data can also be set here. e.g.
-
-/* corgi audio private data */
-static struct wm8731_setup_data corgi_wm8731_setup = {
- .i2c_address = 0x1b,
-};
-
-/* corgi audio subsystem */
-static struct snd_soc_device corgi_snd_devdata = {
- .machine = &snd_soc_corgi,
- .platform = &pxa2xx_soc_platform,
- .codec_dev = &soc_codec_dev_wm8731,
- .codec_data = &corgi_wm8731_setup,
-};
-
-
Machine Power Map
-----------------
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
index 06d835987c6a..d57efad37e0a 100644
--- a/Documentation/sound/alsa/soc/platform.txt
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -20,9 +20,10 @@ struct snd_soc_ops {
int (*trigger)(struct snd_pcm_substream *, int);
};
-The platform driver exports its DMA functionality via struct snd_soc_platform:-
+The platform driver exports its DMA functionality via struct
+snd_soc_platform_driver:-
-struct snd_soc_platform {
+struct snd_soc_platform_driver {
char *name;
int (*probe)(struct platform_device *pdev);
@@ -34,6 +35,13 @@ struct snd_soc_platform {
int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
void (*pcm_free)(struct snd_pcm *);
+ /*
+ * For platform caused delay reporting.
+ * Optional.
+ */
+ snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+
/* platform stream ops */
struct snd_pcm_ops *pcm_ops;
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index fa42be529a73..61e36efbf279 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -168,6 +168,9 @@ config SND_SOC_L3
config SND_SOC_DA7210
tristate
+config SND_SOC_DMIC
+ tristate
+
config SND_SOC_MAX98088
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 76304d478912..333910a9f8fb 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -14,6 +14,7 @@ snd-soc-cs42l51-objs := cs42l51.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
+snd-soc-dmic-objs := dmic.o
snd-soc-l3-objs := l3.o
snd-soc-max98088-objs := max98088.o
snd-soc-pcm3008-objs := pcm3008.o
@@ -93,6 +94,7 @@ obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
+obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 3a582caa6ef9..8b51245f2318 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -106,6 +106,21 @@
#define CS4270_MUTE_DAC_A 0x01
#define CS4270_MUTE_DAC_B 0x02
+/* Power-on default values for the registers
+ *
+ * This array contains the power-on default values of the registers, with the
+ * exception of the "CHIPID" register (01h). The lower four bits of that
+ * register contain the hardware revision, so it is treated as volatile.
+ *
+ * Also note that on the CS4270, the first readable register is 1, but ASoC
+ * assumes the first register is 0. Therfore, the array must have an entry for
+ * register 0, but we use cs4270_reg_is_readable() to tell ASoC that it can't
+ * be read.
+ */
+static const u8 cs4270_default_reg_cache[CS4270_LASTREG + 1] = {
+ 0x00, 0x00, 0x00, 0x30, 0x00, 0x60, 0x20, 0x00, 0x00
+};
+
static const char *supply_names[] = {
"va", "vd", "vlc"
};
@@ -178,6 +193,20 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = {
/* The number of MCLK/LRCK ratios supported by the CS4270 */
#define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios)
+static int cs4270_reg_is_readable(unsigned int reg)
+{
+ return (reg >= CS4270_FIRSTREG) && (reg <= CS4270_LASTREG);
+}
+
+static int cs4270_reg_is_volatile(unsigned int reg)
+{
+ /* Unreadable registers are considered volatile */
+ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
+ return 1;
+
+ return reg == CS4270_CHIPID;
+}
+
/**
* cs4270_set_dai_sysclk - determine the CS4270 samples rates.
* @codec_dai: the codec DAI
@@ -263,97 +292,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
/**
- * cs4270_fill_cache - pre-fill the CS4270 register cache.
- * @codec: the codec for this CS4270
- *
- * This function fills in the CS4270 register cache by reading the register
- * values from the hardware.
- *
- * This CS4270 registers are cached to avoid excessive I2C I/O operations.
- * After the initial read to pre-fill the cache, the CS4270 never updates
- * the register values, so we won't have a cache coherency problem.
- *
- * We use the auto-increment feature of the CS4270 to read all registers in
- * one shot.
- */
-static int cs4270_fill_cache(struct snd_soc_codec *codec)
-{
- u8 *cache = codec->reg_cache;
- struct i2c_client *i2c_client = codec->control_data;
- s32 length;
-
- length = i2c_smbus_read_i2c_block_data(i2c_client,
- CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache);
-
- if (length != CS4270_NUMREGS) {
- dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
- i2c_client->addr);
- return -EIO;
- }
-
- return 0;
-}
-
-/**
- * cs4270_read_reg_cache - read from the CS4270 register cache.
- * @codec: the codec for this CS4270
- * @reg: the register to read
- *
- * This function returns the value for a given register. It reads only from
- * the register cache, not the hardware itself.
- *
- * This CS4270 registers are cached to avoid excessive I2C I/O operations.
- * After the initial read to pre-fill the cache, the CS4270 never updates
- * the register values, so we won't have a cache coherency problem.
- */
-static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u8 *cache = codec->reg_cache;
-
- if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
- return -EIO;
-
- return cache[reg - CS4270_FIRSTREG];
-}
-
-/**
- * cs4270_i2c_write - write to a CS4270 register via the I2C bus.
- * @codec: the codec for this CS4270
- * @reg: the register to write
- * @value: the value to write to the register
- *
- * This function writes the given value to the given CS4270 register, and
- * also updates the register cache.
- *
- * Note that we don't use the hw_write function pointer of snd_soc_codec.
- * That's because it's too clunky: the hw_write_t prototype does not match
- * i2c_smbus_write_byte_data(), and it's just another layer of overhead.
- */
-static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 *cache = codec->reg_cache;
-
- if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
- return -EIO;
-
- /* Only perform an I2C operation if the new value is different */
- if (cache[reg - CS4270_FIRSTREG] != value) {
- struct i2c_client *client = codec->control_data;
- if (i2c_smbus_write_byte_data(client, reg, value)) {
- dev_err(codec->dev, "i2c write failed\n");
- return -EIO;
- }
-
- /* We've written to the hardware, so update the cache */
- cache[reg - CS4270_FIRSTREG] = value;
- }
-
- return 0;
-}
-
-/**
* cs4270_hw_params - program the CS4270 with the given hardware parameters.
* @substream: the audio stream
* @params: the hardware parameters to set
@@ -550,15 +488,16 @@ static struct snd_soc_dai_driver cs4270_dai = {
static int cs4270_probe(struct snd_soc_codec *codec)
{
struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
- int i, ret, reg;
+ int i, ret;
codec->control_data = cs4270->control_data;
- /* The I2C interface is set up, so pre-fill our register cache */
-
- ret = cs4270_fill_cache(codec);
+ /* 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, cs4270->control_type);
if (ret < 0) {
- dev_err(codec->dev, "failed to fill register cache\n");
+ dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret);
return ret;
}
@@ -567,10 +506,7 @@ static int cs4270_probe(struct snd_soc_codec *codec)
* this feature disabled by default. An application (e.g. alsactl) can
* re-enabled it by using the controls.
*/
-
- reg = cs4270_read_reg_cache(codec, CS4270_MUTE);
- reg &= ~CS4270_MUTE_AUTO;
- ret = cs4270_i2c_write(codec, CS4270_MUTE, reg);
+ ret = snd_soc_update_bits(codec, CS4270_MUTE, CS4270_MUTE_AUTO, 0);
if (ret < 0) {
dev_err(codec->dev, "i2c write failed\n");
return ret;
@@ -581,10 +517,8 @@ static int cs4270_probe(struct snd_soc_codec *codec)
* playback has started. An application (e.g. alsactl) can
* re-enabled it by using the controls.
*/
-
- reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
- reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
- ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
+ ret = snd_soc_update_bits(codec, CS4270_TRANS,
+ CS4270_TRANS_SOFT | CS4270_TRANS_ZERO, 0);
if (ret < 0) {
dev_err(codec->dev, "i2c write failed\n");
return ret;
@@ -707,15 +641,16 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec)
* Assign this variable to the codec_dev field of the machine driver's
* snd_soc_device structure.
*/
-static struct snd_soc_codec_driver soc_codec_device_cs4270 = {
- .probe = cs4270_probe,
- .remove = cs4270_remove,
- .suspend = cs4270_soc_suspend,
- .resume = cs4270_soc_resume,
- .read = cs4270_read_reg_cache,
- .write = cs4270_i2c_write,
- .reg_cache_size = CS4270_NUMREGS,
- .reg_word_size = sizeof(u8),
+static const struct snd_soc_codec_driver soc_codec_device_cs4270 = {
+ .probe = cs4270_probe,
+ .remove = cs4270_remove,
+ .suspend = cs4270_soc_suspend,
+ .resume = cs4270_soc_resume,
+ .volatile_register = cs4270_reg_is_volatile,
+ .readable_register = cs4270_reg_is_readable,
+ .reg_cache_size = CS4270_LASTREG + 1,
+ .reg_word_size = sizeof(u8),
+ .reg_cache_default = cs4270_default_reg_cache,
};
/**
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
new file mode 100644
index 000000000000..57e9dac88d38
--- /dev/null
+++ b/sound/soc/codecs/dmic.c
@@ -0,0 +1,81 @@
+/*
+ * dmic.c -- SoC audio for Generic Digital MICs
+ *
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+static struct snd_soc_dai_driver dmic_dai = {
+ .name = "dmic-hifi",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE
+ | SNDRV_PCM_FMTBIT_S24_LE
+ | SNDRV_PCM_FMTBIT_S16_LE,
+ },
+};
+
+static struct snd_soc_codec_driver soc_dmic = {};
+
+static int __devinit dmic_dev_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev,
+ &soc_dmic, &dmic_dai, 1);
+}
+
+static int __devexit dmic_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+MODULE_ALIAS("platform:dmic-codec");
+
+static struct platform_driver dmic_driver = {
+ .driver = {
+ .name = "dmic-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = dmic_dev_probe,
+ .remove = __devexit_p(dmic_dev_remove),
+};
+
+static int __init dmic_init(void)
+{
+ return platform_driver_register(&dmic_driver);
+}
+module_init(dmic_init);
+
+static void __exit dmic_exit(void)
+{
+ platform_driver_unregister(&dmic_driver);
+}
+module_exit(dmic_exit);
+
+MODULE_DESCRIPTION("Generic DMIC driver");
+MODULE_AUTHOR("Liam Girdwood <lrg@slimlogic.co.uk>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 776ac80cc1a8..71d7be8ac488 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -42,14 +42,15 @@
#include <sound/tlv320dac33-plat.h>
#include "tlv320dac33.h"
-#define DAC33_BUFFER_SIZE_BYTES 24576 /* bytes, 12288 16 bit words,
- * 6144 stereo */
-#define DAC33_BUFFER_SIZE_SAMPLES 6144
-
-#define NSAMPLE_MAX 5700
-
-#define MODE7_LTHR 10
-#define MODE7_UTHR (DAC33_BUFFER_SIZE_SAMPLES - 10)
+/*
+ * The internal FIFO is 24576 bytes long
+ * It can be configured to hold 16bit or 24bit samples
+ * In 16bit configuration the FIFO can hold 6144 stereo samples
+ * In 24bit configuration the FIFO can hold 4096 stereo samples
+ */
+#define DAC33_FIFO_SIZE_16BIT 6144
+#define DAC33_FIFO_SIZE_24BIT 4096
+#define DAC33_MODE7_MARGIN 10 /* Safety margin for FIFO in Mode7 */
#define BURST_BASEFREQ_HZ 49152000
@@ -99,16 +100,11 @@ struct tlv320dac33_priv {
unsigned int refclk;
unsigned int alarm_threshold; /* set to be half of LATENCY_TIME_MS */
- unsigned int nsample_min; /* nsample should not be lower than
- * this */
- unsigned int nsample_max; /* nsample should not be higher than
- * this */
enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */
+ unsigned int fifo_size; /* Size of the FIFO in samples */
unsigned int nsample; /* burst read amount from host */
int mode1_latency; /* latency caused by the i2c writes in
* us */
- int auto_fifo_config; /* Configure the FIFO based on the
- * period size */
u8 burst_bclkdiv; /* BCLK divider value in burst mode */
unsigned int burst_rate; /* Interface speed in Burst modes */
@@ -302,7 +298,6 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
if (unlikely(!dac33->chip_power))
return;
- /* 44-46: DAC Control Registers */
/* A : DAC sample rate Fsref/1.5 */
dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0));
/* B : DAC src=normal, not muted */
@@ -325,6 +320,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
dac33_read_reg_cache(codec, DAC33_LINEL_TO_LLO_VOL));
dac33_write(codec, DAC33_LINER_TO_RLO_VOL,
dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL));
+
+ dac33_write(codec, DAC33_OUT_AMP_CTRL,
+ dac33_read_reg_cache(codec, DAC33_OUT_AMP_CTRL));
+
}
static inline int dac33_read_id(struct snd_soc_codec *codec)
@@ -436,73 +435,6 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static int dac33_get_nsample(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-
- ucontrol->value.integer.value[0] = dac33->nsample;
-
- return 0;
-}
-
-static int dac33_set_nsample(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
- int ret = 0;
-
- if (dac33->nsample == ucontrol->value.integer.value[0])
- return 0;
-
- if (ucontrol->value.integer.value[0] < dac33->nsample_min ||
- ucontrol->value.integer.value[0] > dac33->nsample_max) {
- ret = -EINVAL;
- } else {
- dac33->nsample = ucontrol->value.integer.value[0];
- /* Re calculate the burst time */
- dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate,
- dac33->nsample);
- }
-
- return ret;
-}
-
-static int dac33_get_uthr(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-
- ucontrol->value.integer.value[0] = dac33->uthr;
-
- return 0;
-}
-
-static int dac33_set_uthr(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
- int ret = 0;
-
- if (dac33->substream)
- return -EBUSY;
-
- if (dac33->uthr == ucontrol->value.integer.value[0])
- return 0;
-
- if (ucontrol->value.integer.value[0] < (MODE7_LTHR + 10) ||
- ucontrol->value.integer.value[0] > MODE7_UTHR)
- ret = -EINVAL;
- else
- dac33->uthr = ucontrol->value.integer.value[0];
-
- return ret;
-}
-
static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -587,13 +519,6 @@ static const struct snd_kcontrol_new dac33_mode_snd_controls[] = {
dac33_get_fifo_mode, dac33_set_fifo_mode),
};
-static const struct snd_kcontrol_new dac33_fifo_snd_controls[] = {
- SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
- dac33_get_nsample, dac33_set_nsample),
- SOC_SINGLE_EXT("UTHR", 0, 0, MODE7_UTHR, 0,
- dac33_get_uthr, dac33_set_uthr),
-};
-
/* Analog bypass */
static const struct snd_kcontrol_new dac33_dapm_abypassl_control =
SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1);
@@ -601,6 +526,25 @@ static const struct snd_kcontrol_new dac33_dapm_abypassl_control =
static const struct snd_kcontrol_new dac33_dapm_abypassr_control =
SOC_DAPM_SINGLE("Switch", DAC33_LINER_TO_RLO_VOL, 7, 1, 1);
+/* LOP L/R invert selection */
+static const char *dac33_lr_lom_texts[] = {"DAC", "LOP"};
+
+static const struct soc_enum dac33_left_lom_enum =
+ SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 3,
+ ARRAY_SIZE(dac33_lr_lom_texts),
+ dac33_lr_lom_texts);
+
+static const struct snd_kcontrol_new dac33_dapm_left_lom_control =
+SOC_DAPM_ENUM("Route", dac33_left_lom_enum);
+
+static const struct soc_enum dac33_right_lom_enum =
+ SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 2,
+ ARRAY_SIZE(dac33_lr_lom_texts),
+ dac33_lr_lom_texts);
+
+static const struct snd_kcontrol_new dac33_dapm_right_lom_control =
+SOC_DAPM_ENUM("Route", dac33_right_lom_enum);
+
static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("LEFT_LO"),
SND_SOC_DAPM_OUTPUT("RIGHT_LO"),
@@ -617,6 +561,18 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = {
SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0,
&dac33_dapm_abypassr_control),
+ SND_SOC_DAPM_MUX("Left LOM Inverted From", SND_SOC_NOPM, 0, 0,
+ &dac33_dapm_left_lom_control),
+ SND_SOC_DAPM_MUX("Right LOM Inverted From", SND_SOC_NOPM, 0, 0,
+ &dac33_dapm_right_lom_control),
+ /*
+ * For DAPM path, when only the anlog bypass path is enabled, and the
+ * LOP inverted from the corresponding DAC side.
+ * This is needed, so we can attach the DAC power supply in this case.
+ */
+ SND_SOC_DAPM_PGA("Left Bypass PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Bypass PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+
SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amplifier",
DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0),
SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amplifier",
@@ -639,11 +595,22 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Output Left Amplifier", NULL, "DACL"},
{"Output Right Amplifier", NULL, "DACR"},
- {"Output Left Amplifier", NULL, "Analog Left Bypass"},
- {"Output Right Amplifier", NULL, "Analog Right Bypass"},
+ {"Left Bypass PGA", NULL, "Analog Left Bypass"},
+ {"Right Bypass PGA", NULL, "Analog Right Bypass"},
- {"Output Left Amplifier", NULL, "Left DAC Power"},
- {"Output Right Amplifier", NULL, "Right DAC Power"},
+ {"Left LOM Inverted From", "DAC", "Left Bypass PGA"},
+ {"Right LOM Inverted From", "DAC", "Right Bypass PGA"},
+ {"Left LOM Inverted From", "LOP", "Analog Left Bypass"},
+ {"Right LOM Inverted From", "LOP", "Analog Right Bypass"},
+
+ {"Output Left Amplifier", NULL, "Left LOM Inverted From"},
+ {"Output Right Amplifier", NULL, "Right LOM Inverted From"},
+
+ {"DACL", NULL, "Left DAC Power"},
+ {"DACR", NULL, "Right DAC Power"},
+
+ {"Left Bypass PGA", NULL, "Left DAC Power"},
+ {"Right Bypass PGA", NULL, "Right DAC Power"},
/* output */
{"LEFT_LO", NULL, "Output Left Amplifier"},
@@ -732,7 +699,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
spin_unlock_irq(&dac33->lock);
dac33_write16(codec, DAC33_PREFILL_MSB,
- DAC33_THRREG(MODE7_LTHR));
+ DAC33_THRREG(DAC33_MODE7_MARGIN));
/* Enable Upper Threshold IRQ */
dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MUT);
@@ -842,6 +809,8 @@ static int dac33_startup(struct snd_pcm_substream *substream,
/* Stream started, save the substream pointer */
dac33->substream = substream;
+ snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24);
+
return 0;
}
@@ -853,18 +822,17 @@ static void dac33_shutdown(struct snd_pcm_substream *substream,
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
dac33->substream = NULL;
-
- /* Reset the nSample restrictions */
- dac33->nsample_min = 0;
- dac33->nsample_max = NSAMPLE_MAX;
}
+#define CALC_BURST_RATE(bclkdiv, bclk_per_sample) \
+ (BURST_BASEFREQ_HZ / bclkdiv / bclk_per_sample)
static int dac33_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 tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
/* Check parameters for validity */
switch (params_rate(params)) {
@@ -879,6 +847,12 @@ static int dac33_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
+ dac33->fifo_size = DAC33_FIFO_SIZE_16BIT;
+ dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 32);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ dac33->fifo_size = DAC33_FIFO_SIZE_24BIT;
+ dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 64);
break;
default:
dev_err(codec->dev, "unsupported format %d\n",
@@ -933,6 +907,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
aictrl_a |= (DAC33_NCYCL_16 | DAC33_WLEN_16);
fifoctrl_a |= DAC33_WIDTH;
break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ aictrl_a |= (DAC33_NCYCL_32 | DAC33_WLEN_24);
+ break;
default:
dev_err(codec->dev, "unsupported format %d\n",
substream->runtime->format);
@@ -1067,7 +1044,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C,
dac33->burst_bclkdiv);
else
- dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32);
+ if (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE)
+ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32);
+ else
+ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 16);
switch (dac33->fifo_mode) {
case DAC33_FIFO_MODE1:
@@ -1080,7 +1060,8 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
* at the bottom, and also at the top of the FIFO
*/
dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(dac33->uthr));
- dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR));
+ dac33_write16(codec, DAC33_LTHR_MSB,
+ DAC33_THRREG(DAC33_MODE7_MARGIN));
break;
default:
break;
@@ -1109,42 +1090,21 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
/* Number of samples under i2c latency */
dac33->alarm_threshold = US_TO_SAMPLES(rate,
dac33->mode1_latency);
- nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
- dac33->alarm_threshold;
-
- if (dac33->auto_fifo_config) {
- if (period_size <= dac33->alarm_threshold)
- /*
- * Configure nSamaple to number of periods,
- * which covers the latency requironment.
- */
- dac33->nsample = period_size *
- ((dac33->alarm_threshold / period_size) +
- (dac33->alarm_threshold % period_size ?
- 1 : 0));
- else if (period_size > nsample_limit)
- dac33->nsample = nsample_limit;
- else
- dac33->nsample = period_size;
- } else {
- /* nSample time shall not be shorter than i2c latency */
- dac33->nsample_min = dac33->alarm_threshold;
+ nsample_limit = dac33->fifo_size - dac33->alarm_threshold;
+
+ if (period_size <= dac33->alarm_threshold)
/*
- * nSample should not be bigger than alsa buffer minus
- * size of one period to avoid overruns
+ * Configure nSamaple to number of periods,
+ * which covers the latency requironment.
*/
- dac33->nsample_max = substream->runtime->buffer_size -
- period_size;
-
- if (dac33->nsample_max > nsample_limit)
- dac33->nsample_max = nsample_limit;
-
- /* Correct the nSample if it is outside of the ranges */
- if (dac33->nsample < dac33->nsample_min)
- dac33->nsample = dac33->nsample_min;
- if (dac33->nsample > dac33->nsample_max)
- dac33->nsample = dac33->nsample_max;
- }
+ dac33->nsample = period_size *
+ ((dac33->alarm_threshold / period_size) +
+ (dac33->alarm_threshold % period_size ?
+ 1 : 0));
+ else if (period_size > nsample_limit)
+ dac33->nsample = nsample_limit;
+ else
+ dac33->nsample = period_size;
dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate,
dac33->nsample);
@@ -1152,19 +1112,16 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
dac33->t_stamp2 = 0;
break;
case DAC33_FIFO_MODE7:
- if (dac33->auto_fifo_config) {
- dac33->uthr = UTHR_FROM_PERIOD_SIZE(
- period_size,
- rate,
- dac33->burst_rate) + 9;
- if (dac33->uthr > MODE7_UTHR)
- dac33->uthr = MODE7_UTHR;
- if (dac33->uthr < (MODE7_LTHR + 10))
- dac33->uthr = (MODE7_LTHR + 10);
- }
+ dac33->uthr = UTHR_FROM_PERIOD_SIZE(period_size, rate,
+ dac33->burst_rate) + 9;
+ if (dac33->uthr > (dac33->fifo_size - DAC33_MODE7_MARGIN))
+ dac33->uthr = dac33->fifo_size - DAC33_MODE7_MARGIN;
+ if (dac33->uthr < (DAC33_MODE7_MARGIN + 10))
+ dac33->uthr = (DAC33_MODE7_MARGIN + 10);
+
dac33->mode7_us_to_lthr =
SAMPLES_TO_US(substream->runtime->rate,
- dac33->uthr - MODE7_LTHR + 1);
+ dac33->uthr - DAC33_MODE7_MARGIN + 1);
dac33->t_stamp1 = 0;
break;
default:
@@ -1282,8 +1239,8 @@ static snd_pcm_sframes_t dac33_dai_delay(
samples += (samples_in - samples_out);
if (likely(samples > 0))
- delay = samples > DAC33_BUFFER_SIZE_SAMPLES ?
- DAC33_BUFFER_SIZE_SAMPLES : samples;
+ delay = samples > dac33->fifo_size ?
+ dac33->fifo_size : samples;
else
delay = 0;
}
@@ -1335,7 +1292,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
samples_in = US_TO_SAMPLES(
dac33->burst_rate,
time_delta);
- delay = MODE7_LTHR + samples_in - samples_out;
+ delay = DAC33_MODE7_MARGIN + samples_in - samples_out;
if (unlikely(delay > uthr))
delay = uthr;
@@ -1486,14 +1443,10 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
snd_soc_add_controls(codec, dac33_snd_controls,
ARRAY_SIZE(dac33_snd_controls));
/* Only add the FIFO controls, if we have valid IRQ number */
- if (dac33->irq >= 0) {
+ if (dac33->irq >= 0)
snd_soc_add_controls(codec, dac33_mode_snd_controls,
ARRAY_SIZE(dac33_mode_snd_controls));
- /* FIFO usage controls only, if autoio config is not selected */
- if (!dac33->auto_fifo_config)
- snd_soc_add_controls(codec, dac33_fifo_snd_controls,
- ARRAY_SIZE(dac33_fifo_snd_controls));
- }
+
dac33_add_widgets(codec);
err_power:
@@ -1542,7 +1495,7 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = {
#define DAC33_RATES (SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000)
-#define DAC33_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+#define DAC33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
static struct snd_soc_dai_ops dac33_dai_ops = {
.startup = dac33_startup,
@@ -1590,17 +1543,11 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
dac33->power_gpio = pdata->power_gpio;
dac33->burst_bclkdiv = pdata->burst_bclkdiv;
- /* Pre calculate the burst rate */
- dac33->burst_rate = BURST_BASEFREQ_HZ / dac33->burst_bclkdiv / 32;
dac33->keep_bclk = pdata->keep_bclk;
- dac33->auto_fifo_config = pdata->auto_fifo_config;
dac33->mode1_latency = pdata->mode1_latency;
if (!dac33->mode1_latency)
dac33->mode1_latency = 10000; /* 10ms */
dac33->irq = client->irq;
- dac33->nsample = NSAMPLE_MAX;
- dac33->nsample_max = NSAMPLE_MAX;
- dac33->uthr = MODE7_UTHR;
/* Disable FIFO use by default */
dac33->fifo_mode = DAC33_FIFO_BYPASS;
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 0a99f313e218..1f1ac8110bef 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -339,7 +339,6 @@ EXPORT_SYMBOL_GPL(tpa6130a2_stereo_enable);
int tpa6130a2_add_controls(struct snd_soc_codec *codec)
{
struct tpa6130a2_data *data;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
if (tpa6130a2_client == NULL)
return -ENODEV;
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 2f68f5949a63..4bbf1b15a493 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1138,19 +1138,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
SND_SOC_NOPM, 0, 0, &hsr_mux_controls),
/* Analog playback drivers */
- SND_SOC_DAPM_PGA_E("Handsfree Left Driver",
+ SND_SOC_DAPM_OUT_DRV_E("Handsfree Left Driver",
TWL6040_REG_HFLCTL, 4, 0, NULL, 0,
pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_PGA_E("Handsfree Right Driver",
+ SND_SOC_DAPM_OUT_DRV_E("Handsfree Right Driver",
TWL6040_REG_HFRCTL, 4, 0, NULL, 0,
pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_PGA_E("Headset Left Driver",
+ SND_SOC_DAPM_OUT_DRV_E("Headset Left Driver",
TWL6040_REG_HSLCTL, 2, 0, NULL, 0,
pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_PGA_E("Headset Right Driver",
+ SND_SOC_DAPM_OUT_DRV_E("Headset Right Driver",
TWL6040_REG_HSRCTL, 2, 0, NULL, 0,
pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 96e0dc09f203..ac210ccebd4b 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -30,7 +30,7 @@
#include "wm8995.h"
-static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] __devinitconst = {
+static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] = {
[0] = 0x8995, [5] = 0x0100, [16] = 0x000b, [17] = 0x000b,
[24] = 0x02c0, [25] = 0x02c0, [26] = 0x02c0, [27] = 0x02c0,
[28] = 0x000f, [32] = 0x0005, [33] = 0x0005, [40] = 0x0003,
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 7e84f24b9a88..d203f4da18a0 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -102,6 +102,17 @@ static const int omap24xx_dma_reqs[][2] = {
static const int omap24xx_dma_reqs[][2] = {};
#endif
+#if defined(CONFIG_ARCH_OMAP4)
+static const int omap44xx_dma_reqs[][2] = {
+ { OMAP44XX_DMA_MCBSP1_TX, OMAP44XX_DMA_MCBSP1_RX },
+ { OMAP44XX_DMA_MCBSP2_TX, OMAP44XX_DMA_MCBSP2_RX },
+ { OMAP44XX_DMA_MCBSP3_TX, OMAP44XX_DMA_MCBSP3_RX },
+ { OMAP44XX_DMA_MCBSP4_TX, OMAP44XX_DMA_MCBSP4_RX },
+};
+#else
+static const int omap44xx_dma_reqs[][2] = {};
+#endif
+
#if defined(CONFIG_ARCH_OMAP2420)
static const unsigned long omap2420_mcbsp_port[][2] = {
{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
@@ -147,6 +158,21 @@ static const unsigned long omap34xx_mcbsp_port[][2] = {
static const unsigned long omap34xx_mcbsp_port[][2] = {};
#endif
+#if defined(CONFIG_ARCH_OMAP4)
+static const unsigned long omap44xx_mcbsp_port[][2] = {
+ { OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
+};
+#else
+static const unsigned long omap44xx_mcbsp_port[][2] = {};
+#endif
+
static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -224,7 +250,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
* 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
* 4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
*/
- if (cpu_is_omap343x()) {
+ if (cpu_is_omap343x() || cpu_is_omap44xx()) {
/*
* Rule for the buffer size. We should not allow
* smaller buffer than the FIFO size to avoid underruns
@@ -332,6 +358,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
} else if (cpu_is_omap343x()) {
dma = omap24xx_dma_reqs[bus_id][substream->stream];
port = omap34xx_mcbsp_port[bus_id][substream->stream];
+ } else if (cpu_is_omap44xx()) {
+ dma = omap44xx_dma_reqs[bus_id][substream->stream];
+ port = omap44xx_mcbsp_port[bus_id][substream->stream];
} else {
return -ENODEV;
}
@@ -498,11 +527,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
regs->spcr2 |= XINTM(3) | FREE;
regs->spcr1 |= RINTM(3);
/* RFIG and XFIG are not defined in 34xx */
- if (!cpu_is_omap34xx()) {
+ if (!cpu_is_omap34xx() && !cpu_is_omap44xx()) {
regs->rcr2 |= RFIG;
regs->xcr2 |= XFIG;
}
- if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+ if (cpu_is_omap2430() || cpu_is_omap34xx() || cpu_is_omap44xx()) {
regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE;
regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE;
}
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index ffdcc5abb7b9..110c106611d3 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -50,6 +50,10 @@ enum omap_mcbsp_div {
#undef NUM_LINKS
#define NUM_LINKS 3
#endif
+#if defined(CONFIG_ARCH_OMAP4)
+#undef NUM_LINKS
+#define NUM_LINKS 4
+#endif
#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3)
#undef NUM_LINKS
#define NUM_LINKS 5
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 5f2479ce9dde..1e574a5d440d 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -235,6 +235,7 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
snd_soc_dapm_sync(dapm);
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index df13338cb3e2..6088a6a3238a 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -8,11 +8,11 @@
* published by the Free Software Foundation.
*/
+#include <linux/clkdev.h>
#include <linux/device.h>
#include <linux/firmware.h>
#include <linux/module.h>
-#include <asm/clkdev.h>
#include <asm/clock.h>
#include <cpu/sh7722.h>
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index b2e333f5a388..1a36b36c5baa 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -1339,7 +1339,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
goto err;
}
lzo_blocks[i]->sync_bmp = sync_bmp;
- lzo_blocks[i]->sync_bmp_nbits = reg_size;
+ lzo_blocks[i]->sync_bmp_nbits = bmp_size;
/* alloc the working space for the compressed block */
ret = snd_soc_lzo_prepare(lzo_blocks[i]);
if (ret < 0)
generated by cgit v1.2.3 (git 2.39.0) at 2024-12-30 13:37:26 +0300