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author | Takashi Iwai <tiwai@suse.de> | 2011-01-12 07:20:30 +0100 |
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committer | Takashi Iwai <tiwai@suse.de> | 2011-01-12 07:20:30 +0100 |
commit | 2f55eaab56ea083498034c18e09d356594081fca (patch) | |
tree | 41563dd4dfe981514915dfcf57e5a5b298c82b6a /sound | |
parent | ASoC: Fix section mismatch in wm8995.c (diff) | |
parent | ASoC: tpa6130a2: Fix compiler warning (diff) | |
download | linux-2f55eaab56ea083498034c18e09d356594081fca.tar.xz linux-2f55eaab56ea083498034c18e09d356594081fca.zip |
Merge branch 'for-2.6.38' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
Diffstat (limited to 'sound')
-rw-r--r-- | sound/soc/codecs/Kconfig | 3 | ||||
-rw-r--r-- | sound/soc/codecs/Makefile | 2 | ||||
-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/omap/omap-mcbsp.c | 35 | ||||
-rw-r--r-- | sound/soc/omap/omap-mcbsp.h | 4 |
8 files changed, 234 insertions, 169 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 054191ed08ef..883a312bb293 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -167,6 +167,9 @@ 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 6a1e17bbbf83..579af9c4f128 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 @@ -92,6 +93,7 @@ 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/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/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 |