diff options
| -rw-r--r-- | Documentation/sound/alsa/soc/codec.txt | 45 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/machine.txt | 38 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/platform.txt | 12 | ||||
| -rw-r--r-- | sound/soc/codecs/Kconfig | 3 | ||||
| -rw-r--r-- | sound/soc/codecs/Makefile | 2 | ||||
| -rw-r--r-- | sound/soc/codecs/cs4270.c | 161 | ||||
| -rw-r--r-- | sound/soc/codecs/dmic.c | 81 | ||||
| -rw-r--r-- | sound/soc/codecs/tlv320dac33.c | 269 | ||||
| -rw-r--r-- | sound/soc/codecs/tpa6130a2.c | 1 | ||||
| -rw-r--r-- | sound/soc/codecs/twl6040.c | 8 | ||||
| -rw-r--r-- | sound/soc/codecs/wm8995.c | 2 | ||||
| -rw-r--r-- | sound/soc/omap/omap-mcbsp.c | 35 | ||||
| -rw-r--r-- | sound/soc/omap/omap-mcbsp.h | 4 | ||||
| -rw-r--r-- | sound/soc/samsung/rx1950_uda1380.c | 1 | ||||
| -rw-r--r-- | sound/soc/sh/migor.c | 2 | ||||
| -rw-r--r-- | sound/soc/soc-cache.c | 2 |
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) |