March 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: omap: rx51: Enable McBSP2 sidetone
by Jarkko Nikula 09 Mar '11

09 Mar '11

McBSP sidetone is needed in telephony applications. McBSP sidetone is a configurable FIR filter that forms a loopback from McBSP input to output. This patch enables the McBSP2 sidetone ALSA controls so that it can be used on Nokia RX-51/N900. Sidetone feature can be tested with following commands: (set up codec input and output paths) # Enable and configure sidetone amixer -D hw:0 set 'McBSP2 Sidetone' on amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767 echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps # Do not loop audio via CPU arecord -f dat >/dev/null |aplay /dev/zero Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/omap/rx51.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 055e447..d098622 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -331,6 +331,10 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + err = omap_mcbsp_st_add_controls(codec, 1); + if (err < 0) + return err; + snd_soc_dapm_sync(dapm); /* AV jack detection */ -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: Use the correct DAPM context when cleaning up final widget set
by Mark Brown 09 Mar '11

09 Mar '11

Now we've got multi-component we need to make sure that the DAPM context (and hence register I/O context) we use to apply the pending updates at the end of a DAPM sequence is the one we were processing rather than the one that was used to initate the state change. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a6fb85d..c5335da 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -984,7 +984,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, } if (!list_empty(&pending)) - dapm_seq_run_coalesced(dapm, &pending); + dapm_seq_run_coalesced(cur_dapm, &pending); if (cur_dapm && cur_dapm->seq_notifier) { for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Simplify WM9081 speaker startup by using widgets for spaker output
by Mark Brown 09 Mar '11

09 Mar '11

Now we have a register write minimisation code in DAPM we don't need to worry about the ordering of the enable and disable of the PGA and the output stage. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 32 ++++++-------------------------- 1 files changed, 6 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index effaf75..55cdf29 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -389,27 +389,6 @@ SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0), SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0), }; -static int speaker_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - unsigned int reg = snd_soc_read(codec, WM9081_POWER_MANAGEMENT); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - reg |= WM9081_SPK_ENA; - break; - - case SND_SOC_DAPM_PRE_PMD: - reg &= ~WM9081_SPK_ENA; - break; - } - - snd_soc_write(codec, WM9081_POWER_MANAGEMENT, reg); - - return 0; -} - struct _fll_div { u16 fll_fratio; u16 fll_outdiv; @@ -747,9 +726,8 @@ SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0), -SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0, - speaker_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LINEOUT"), SND_SOC_DAPM_OUTPUT("SPKN"), @@ -780,8 +758,10 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "Speaker PGA", NULL, "TOCLK" }, { "Speaker PGA", NULL, "CLK_SYS" }, - { "SPKN", NULL, "Speaker PGA" }, - { "SPKP", NULL, "Speaker PGA" }, + { "Speaker", NULL, "Speaker PGA" }, + + { "SPKN", NULL, "Speaker" }, + { "SPKP", NULL, "Speaker" }, }; static int wm9081_set_bias_level(struct snd_soc_codec *codec, -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew()
by Mark Brown 09 Mar '11

09 Mar '11

Currently will ignore prefixes when creating DAPM controls. Since currently all control creation goes through snd_soc_cnew() we can fix this by factoring the prefixing into that function. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 3 ++- sound/soc/soc-core.c | 45 +++++++++++++++++++++++++++++++-------------- sound/soc/soc-dapm.c | 16 ++++++++++++++-- 3 files changed, 47 insertions(+), 17 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8b097f7..876dd5a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -341,7 +341,8 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); *Controls */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, - void *data, char *long_name); + void *data, char *long_name, + const char *prefix); int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index db3075d..17efacd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2344,22 +2344,45 @@ EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); * @_template: control template * @data: control private data * @long_name: control long name + * @prefix: control name prefix * * Create a new mixer control from a template control. * * Returns 0 for success, else error. */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, - void *data, char *long_name) + void *data, char *long_name, + const char *prefix) { struct snd_kcontrol_new template; + struct snd_kcontrol *kcontrol; + char *name = NULL; + int name_len; memcpy(&template, _template, sizeof(template)); - if (long_name) - template.name = long_name; template.index = 0; - return snd_ctl_new1(&template, data); + if (!long_name) + long_name = template.name; + + if (prefix) { + name_len = strlen(long_name) + strlen(prefix) + 2; + name = kmalloc(name_len, GFP_ATOMIC); + if (!name) + return NULL; + + snprintf(name, name_len, "%s %s", prefix, long_name); + + template.name = name; + } else { + template.name = long_name; + } + + kcontrol = snd_ctl_new1(&template, data); + + kfree(name); + + return kcontrol; } EXPORT_SYMBOL_GPL(snd_soc_cnew); @@ -2378,22 +2401,16 @@ int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = codec->card->snd_card; - char prefixed_name[44], *name; int err, i; for (i = 0; i < num_controls; i++) { const struct snd_kcontrol_new *control = &controls[i]; - if (codec->name_prefix) { - snprintf(prefixed_name, sizeof(prefixed_name), "%s %s", - codec->name_prefix, control->name); - name = prefixed_name; - } else { - name = control->name; - } - err = snd_ctl_add(card, snd_soc_cnew(control, codec, name)); + err = snd_ctl_add(card, snd_soc_cnew(control, codec, + control->name, + codec->name_prefix)); if (err < 0) { dev_err(codec->dev, "%s: Failed to add %s: %d\n", - codec->name, name, err); + codec->name, control->name, err); return err; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 570db88..a6fb85d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -369,6 +369,12 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, size_t name_len; struct snd_soc_dapm_path *path; struct snd_card *card = dapm->card->snd_card; + const char *prefix; + + if (dapm->codec) + prefix = dapm->codec->name_prefix; + else + prefix = NULL; /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -409,7 +415,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, path->long_name[name_len - 1] = '\0'; path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, - path->long_name); + path->long_name, prefix); ret = snd_ctl_add(card, path->kcontrol); if (ret < 0) { dev_err(dapm->dev, @@ -431,6 +437,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; + const char *prefix; int ret = 0; if (!w->num_kcontrols) { @@ -438,7 +445,12 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, return -EINVAL; } - kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); + if (dapm->codec) + prefix = dapm->codec->name_prefix; + else + prefix = NULL; + + kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name, prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) -- 1.7.2.3

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[PATCH] ALSA: HDA: Remove over-initialization of mux in ALC269VB
by David Henningsson 09 Mar '11

09 Mar '11

This initialization verb breaks the automatic mux setup in some cases (e g just one internal mic and nothing else), and it's not needed as both laptop_amic and laptop_dmic models set this verb anyway. Cc: stable(a)kernel.org (2.6.38+) BugLink: http://bugs.launchpad.net/bugs/732445 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33346c1..cf66d0f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14574,11 +14574,6 @@ static struct hda_verb alc269vb_init_verbs[] = { {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* FIXME: use Mux-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x22, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* set EAPD */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } -- 1.7.1 --------------080502060804070905010802--

1 0

[alsa-devel] [patch] ASoC: sgtl5000: use after free in ldo_regulator_register()
by Dan Carpenter 08 Mar '11

08 Mar '11

The "ldo" variable was dereferenced after free on the error path. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index b7e97c0..1f7217f 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -875,11 +875,13 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, ldo->dev = regulator_register(&ldo->desc, codec->dev, init_data, ldo); if (IS_ERR(ldo->dev)) { + int ret = PTR_ERR(ldo->dev); + dev_err(codec->dev, "failed to register regulator\n"); kfree(ldo->desc.name); kfree(ldo); - return PTR_ERR(ldo->dev); + return ret; } return 0;

3 2

[alsa-devel] [PATCH 1/2] ASoC: Refactor symmetric_rates check to reduce indentation
by Mark Brown 08 Mar '11

08 Mar '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 28 ++++++++++++++-------------- 1 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c2ec6cb..65ff612 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -500,20 +500,20 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->driver->symmetric_rates || cpu_dai->driver->symmetric_rates || - rtd->dai_link->symmetric_rates) { - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", - rtd->rate); - - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, - rtd->rate); - if (ret < 0) { - dev_err(&rtd->dev, - "Unable to apply rate symmetry constraint: %d\n", ret); - return ret; - } + if (!codec_dai->driver->symmetric_rates && + !cpu_dai->driver->symmetric_rates && + !rtd->dai_link->symmetric_rates) + return 0; + + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + rtd->rate, rtd->rate); + if (ret < 0) { + dev_err(&rtd->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; } return 0; -- 1.7.2.3

2 2

[alsa-devel] [PATCH 1/2] ASoC: Provide CODEC clocking operations and API calls
by Mark Brown 08 Mar '11

08 Mar '11

When multi component systems use DAIless amplifiers which require clocking configuration it is at best hard to use the current clocking API as this requires a DAI even though the device may not even have one. Address this by adding set_sysclk() and set_pll() operations and APIs for CODECs. In order to avoid issues with devices which could be used either with or without DAIs make the DAI variants call through to their CODEC counterparts if there is no DAI specific operation. Converting over entirely would create problems for multi-DAI devices which offer per-DAI clocking setup. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 11 +++++++++++ sound/soc/soc-core.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 57 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 363e3a8..8b097f7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -260,6 +260,11 @@ enum snd_soc_compress_type { SND_SOC_RBTREE_COMPRESSION }; +int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir); +int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out); + int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); int snd_soc_suspend(struct device *dev); @@ -569,6 +574,12 @@ struct snd_soc_codec_driver { const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; + /* codec wide operations */ + int (*set_sysclk)(struct snd_soc_codec *codec, + int clk_id, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out); + /* codec IO */ unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 286a385..9b482dd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3062,12 +3062,34 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, { if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); + else if (dai->codec && dai->codec->driver->set_sysclk) + return dai->codec->driver->set_sysclk(dai->codec, clk_id, + freq, dir); else return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); /** + * snd_soc_codec_set_sysclk - configure CODEC system or master clock. + * @codec: CODEC + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir) +{ + if (codec->driver->set_sysclk) + return codec->driver->set_sysclk(codec, clk_id, freq, dir); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); + +/** * snd_soc_dai_set_clkdiv - configure DAI clock dividers. * @dai: DAI * @div_id: DAI specific clock divider ID @@ -3103,11 +3125,35 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, if (dai->driver && dai->driver->ops->set_pll) return dai->driver->ops->set_pll(dai, pll_id, source, freq_in, freq_out); + else if (dai->codec && dai->codec->driver->set_pll) + return dai->codec->driver->set_pll(dai->codec, pll_id, source, + freq_in, freq_out); else return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); +/* + * snd_soc_codec_set_pll - configure codec PLL. + * @codec: CODEC + * @pll_id: DAI specific PLL ID + * @source: DAI specific source for the PLL + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + if (codec->driver->set_pll) + return codec->driver->set_pll(codec, pll_id, source, + freq_in, freq_out); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); + /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI -- 1.7.2.3

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[alsa-devel] [PATCH 1/2] ASoC: Add DAPM widget and path data to CODEC driver structure
by Mark Brown 08 Mar '11

08 Mar '11

Allow a slight simplification of CODEC drivers by allowing DAPM routes and widgets to be provided in a table. They will be instantiated at the end of CODEC probe. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 12 ++++++++++-- 2 files changed, 16 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a23f5a5..363e3a8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -563,6 +563,12 @@ struct snd_soc_codec_driver { pm_message_t state); int (*resume)(struct snd_soc_codec *); + /* Default DAPM setup, added after probe() is run */ + const struct snd_soc_dapm_widget *dapm_widgets; + int num_dapm_widgets; + const struct snd_soc_dapm_route *dapm_routes; + int num_dapm_routes; + /* codec IO */ unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8926d38..286a385 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1464,6 +1464,7 @@ static int soc_probe_codec(struct snd_soc_card *card, struct snd_soc_codec *codec) { int ret = 0; + const struct snd_soc_codec_driver *driver = codec->driver; codec->card = card; codec->dapm.card = card; @@ -1472,8 +1473,8 @@ static int soc_probe_codec(struct snd_soc_card *card, if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; - if (codec->driver->probe) { - ret = codec->driver->probe(codec); + if (driver->probe) { + ret = driver->probe(codec); if (ret < 0) { dev_err(codec->dev, "asoc: failed to probe CODEC %s: %d\n", @@ -1482,6 +1483,13 @@ static int soc_probe_codec(struct snd_soc_card *card, } } + if (driver->dapm_widgets) + snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, + driver->num_dapm_widgets); + if (driver->dapm_routes) + snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, + driver->num_dapm_routes); + soc_init_codec_debugfs(codec); /* mark codec as probed and add to card codec list */ -- 1.7.2.3

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[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 08 Mar '11

08 Mar '11

The following changes since commit 4a5f7bda8fe9d0ed08ed4c5beb5dc3fa62f09d05: ASoC: Add platform data for WM9081 IRQ pin configuration (2011-03-01 23:30:53 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Alexander Sverdlin (2): ASoC: Extend range of supported sample rates for CS4271 CODEC. ASoC: Remove warnings in ep93xx-i2s.c Axel Lin (2): ASoC: tlv320dac33: add MODULE_DEVICE_TABLE ASoC: Constify i2c_device_id table Dimitris Papastamos (1): ASoC: WM8994: Ensure MICBIAS is provided with a clock Javier Martin (2): ASoC: Add TI tlv320aic32x4 codec support. ASoC: Add machine driver for Visstrim_M10 board. Lars-Peter Clausen (6): ASoC: neo1973_wm8753: Remove scenario management code. ASoC: neo1973_wm8753: Move lm4857 specefic code to its own module ASoC: neo1973_gta02_wm8753: Remove lm4853_{set,get}_state ASoC: neo1973_gta02_wm8753: Use gpio_request_array to request gpios ASoC: Samsung: Merge neo1937_wm8753 and neo1973_gta02_wm8753 sound board driver ASoC: Add driver for the dfbmcs320 bluetooth module Mark Brown (11): ASoC: Fix broken bitfield definitions in WM8978 Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Add TLV information for WM8978 DAC limiter ASoC: Get the card directly from the DAPM context ASoC: Check for a CODEC before dereferencing in DAPM ASoC: Add a per-card DAPM context ASoC: Allow card DAPM widgets and routes to be set up at registration ASoC: Add a late_probe() callback to cards ASoC: Fix outdated API usage in tlv320aic32x4 ASoC: Add missing debugfs conditionals ASoC: Add LM4857 to SND_SOC_ALL_CODECS include/sound/soc.h | 12 + include/sound/tlv320aic32x4.h | 31 ++ sound/soc/codecs/Kconfig | 13 +- sound/soc/codecs/Makefile | 6 + sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 7 +- sound/soc/codecs/dfbmcs320.c | 72 +++ sound/soc/codecs/lm4857.c | 276 +++++++++++ sound/soc/codecs/tlv320aic32x4.c | 794 ++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.h | 143 ++++++ sound/soc/codecs/tlv320dac33.c | 1 + sound/soc/codecs/wm8978.c | 17 +- sound/soc/ep93xx/ep93xx-i2s.c | 8 +- sound/soc/imx/Kconfig | 10 + sound/soc/imx/Makefile | 2 + sound/soc/imx/mx27vis-aic32x4.c | 137 +++++ sound/soc/samsung/Kconfig | 19 +- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/lm4857.h | 32 -- sound/soc/samsung/neo1973_gta02_wm8753.c | 494 ------------------- sound/soc/samsung/neo1973_wm8753.c | 634 +++++++++--------------- sound/soc/soc-core.c | 31 ++ sound/soc/soc-dapm.c | 14 +- 23 files changed, 1795 insertions(+), 962 deletions(-) create mode 100644 include/sound/tlv320aic32x4.h create mode 100644 sound/soc/codecs/dfbmcs320.c create mode 100644 sound/soc/codecs/lm4857.c create mode 100644 sound/soc/codecs/tlv320aic32x4.c create mode 100644 sound/soc/codecs/tlv320aic32x4.h create mode 100644 sound/soc/imx/mx27vis-aic32x4.c delete mode 100644 sound/soc/samsung/lm4857.h delete mode 100644 sound/soc/samsung/neo1973_gta02_wm8753.c

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