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

[alsa-devel] ASoC updates for 2.6.39.
by Mark Brown 01 Feb '11

01 Feb '11

The following changes since commit a211701eb1e4a41bbee22d61f35dba8e55925db4: ASoC: sst_platform porting sst dsp driver interface as per latest in Greg's staging tree (2011-01-19 11:28:11 +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 (3): ASoC: CS4271 codec support ASoC: EDB93xx machine sound driver with CS4271 ASoC: cs4271.c: improve error handling Dimitris Papastamos (5): ASoC: WM8995: Fix incorrect use of snd_soc_update_bits() ASoC: soc-cache: Introduce the cache_bypass option ASoC: soc-cache: Apply the cache_bypass option ASoC: soc-cache: Add trace event for snd_soc_cache_sync() ASoC: soc-core: Ensure codec_reg has fixed length fields Dmitry Eremin-Solenikov (1): ASoC: correct link specifications for corgi, poodle and spitz H Hartley Sweeten (1): ASoC: neo1973_wm8753 audio support does not require scoop Harsha Priya (3): ASoC: sn95031: add capture support ASoC: sst_platform: add support for capture stream on headset dai ASoC: mid-x86: Add support for capture in machine driver Janusz Krzysztofik (2): ASoC: Amstrad Delta: fix const related build error ASoC: CX20442: fix NULL pointer dereference Jarkko Nikula (4): ASoC: Fix module refcount for auxiliary devices ASoC: omap: rx51: Add stereo output support to audio jack ASoC: soc-core: Increment codec and platform driver refcounts before probing ASoC: omap: rx51: Add earphone support Kuninori Morimoto (7): ASoC: ak4642: add SND_SOC_DAIFMT_FORMAT support ASoC: sh: fsi: Add fsi_get_priv_frm_dai function ASoC: sh: fsi: Add snd_soc_dai_set_fmt support ASoC: sh: fsi-hdmi: Add FSI port and HDMI selection ASoC: sh: fsi: move chan_num from fsi_stream to fsi_priv ASoC: sh: fsi: free from NULL pointer of struct sh_fsi_platform_info ASoC: sh: fsi: modify selection method of I2S/PCM/SPDIF format Lars-Peter Clausen (2): ASoC: Fix codec device id format used by some dai_links ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s Manjunathappa, Prakash (1): ASoC: DaVinci: fix kernel panic due to uninitialized platform_data Mark Brown (25): ASoC: Explicitly say if we're powering up or down ASoC: Add support for sequencing within ASoC: Provide per widget type callback when executing DAPM sequences Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Staticise twl6040_hs_jack_report() ASoC: Handle low measured DC offsets for wm_hubs devices Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Fix type for snd_soc_volatile_register() ASoC: Remove controls from sequenced PGA arguments Merge branch 'for-2.6.38' into for-2.6.39 Merge branch 'tegra-arch' into for-2.6.39 ASoC: Staticise non-exported symbols in cs4271 ASoC: Use card rather than soc-audio device to card PM functions ASoC: Replace pdev with card in machine driver probe and remove ASoC: Export card PM callbacks for use in direct registered cards ASoC: Make cache status available via debugfs ASoC: Add subsequence information to seq_notify callbacks Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Add card driver data Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Accept any logical value for WM8962 GPIO set() ASoC: Accept any logical value WM8903 GPIO set() Merge branches 'for-2.6.38' and 'tegra-arch' into for-2.6.39 ASoC: Use snd_pcm_format_width() in snd_soc_params_to_frame_size() Merge branch 'for-2.6.39' of git://git.kernel.org/.../lrg/asoc-2.6 into for-2.6.39 Qiao Zhou (1): ASoC: WM8994: fix wrong value in tristate function Rajashekhara, Sudhakar (1): ASoC: da8xx/omap-l1xx: match codec_name with i2c ids Stephen Warren (12): ASoC: wm8903: Expose GPIOs through gpiolib ARM: tegra: Add Harmony sound platform data type ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw() ASoC: Move card list initialization to snd_soc_register_card ASoC: Tegra: Harmony: Don't use soc-audio platform device ASoC: Tegra: Harmony: Support the internal speaker ASoC: Tegra: Harmony: Fix indentation issue. ASoC: Tegra: Harmony: Use dev_err not pr_err ASoC: Tegra: utils: Don't use global variables ASoC: Tegra: I2S: Use dev_err not pr_err ASoC: Tegra: Harmony: Remove redundant !! ARM: tegra: Add to struct harmony_audio_platform_data arch/arm/mach-shmobile/board-ag5evm.c | 10 - arch/arm/mach-shmobile/board-ap4evb.c | 13 +- arch/arm/mach-shmobile/board-mackerel.c | 13 +- arch/arm/mach-tegra/include/mach/harmony_audio.h | 22 + arch/sh/boards/mach-ecovec24/setup.c | 6 +- arch/sh/boards/mach-se/7724/setup.c | 6 +- include/sound/cs4271.h | 25 + include/sound/sh_fsi.h | 76 +-- include/sound/soc-dapm.h | 16 + include/sound/soc.h | 41 +- include/sound/wm8903.h | 20 +- include/trace/events/asoc.h | 25 + sound/soc/atmel/snd-soc-afeb9260.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 24 + sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs4271.c | 659 ++++++++++++++++++++++ sound/soc/codecs/cx20442.c | 3 + sound/soc/codecs/sn95031.c | 281 +++++++++ sound/soc/codecs/twl6040.c | 4 +- sound/soc/codecs/wm8903.c | 127 ++++- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm_hubs.c | 15 +- sound/soc/davinci/davinci-evm.c | 2 +- sound/soc/ep93xx/Kconfig | 9 + sound/soc/ep93xx/Makefile | 2 + sound/soc/ep93xx/edb93xx.c | 142 +++++ sound/soc/fsl/mpc8610_hpcd.c | 6 +- sound/soc/fsl/p1022_ds.c | 6 +- sound/soc/mid-x86/mfld_machine.c | 3 + sound/soc/mid-x86/sst_platform.c | 6 + sound/soc/omap/Kconfig | 1 + sound/soc/omap/ams-delta.c | 2 - sound/soc/omap/rx51.c | 93 +++- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/raumfeld.c | 4 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/tosa.c | 4 +- sound/soc/pxa/zylonite.c | 9 +- sound/soc/samsung/neo1973_gta02_wm8753.c | 6 +- sound/soc/samsung/neo1973_wm8753.c | 7 +- sound/soc/samsung/s3c24xx_simtec_hermes.c | 4 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/sh/fsi-ak4642.c | 13 +- sound/soc/sh/fsi-da7210.c | 13 +- sound/soc/sh/fsi-hdmi.c | 77 +++- sound/soc/sh/fsi.c | 203 ++++--- sound/soc/soc-cache.c | 50 ++- sound/soc/soc-core.c | 114 +++-- sound/soc/soc-dapm.c | 71 ++- sound/soc/soc-utils.c | 23 +- sound/soc/tegra/harmony.c | 170 +++++-- sound/soc/tegra/tegra_asoc_utils.c | 91 ++-- sound/soc/tegra/tegra_asoc_utils.h | 20 +- sound/soc/tegra/tegra_i2s.c | 2 +- 61 files changed, 2136 insertions(+), 437 deletions(-) create mode 100644 arch/arm/mach-tegra/include/mach/harmony_audio.h create mode 100644 include/sound/cs4271.h create mode 100644 sound/soc/codecs/cs4271.c create mode 100644 sound/soc/ep93xx/edb93xx.c

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 01 Feb '11

01 Feb '11

The following changes since commit a3adfa00e8089aa72826c6ba04bcb18cfceaf0a9: ASoC: correct link specifications for corgi, poodle and spitz (2011-01-25 15:18:42 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Nothing exciting here except for Stephen's mask/val_mask fix which fixes a fairly wide breakage. Janusz Krzysztofik (2): ASoC: Amstrad Delta: fix const related build error ASoC: CX20442: fix NULL pointer dereference Jarkko Nikula (1): ASoC: Fix module refcount for auxiliary devices Manjunathappa, Prakash (1): ASoC: DaVinci: fix kernel panic due to uninitialized platform_data Stephen Warren (1): ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw() sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cx20442.c | 3 +++ sound/soc/omap/ams-delta.c | 2 -- sound/soc/soc-core.c | 3 --- sound/soc/soc-dapm.c | 6 +++--- 5 files changed, 7 insertions(+), 9 deletions(-)

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[alsa-devel] Digital IO and SPDIF format
by Torsten Schenk 01 Feb '11

01 Feb '11

Hello, I'm currenty trying to make digital IO working for snd-usb-6fire. I figured out what the card does but I don't know how things (especially input) should be handled in alsa. Unfortunately it is relatively hard to find documentation on that topic so that I decided to ask the pros. The card can receive the whole spdif subframe with all bits (BMW + at least channel status bit and parity bit, but I guess also the other two). It also does receive two spdif-frames per frame (making it possible to receive f.ex. 96kHz digital audio if 6fire's sample rate is set to 48kHz) This also leads to the fact that the 6fire does not drop spdif frames if playback device's sample rate is not very much higher than 6fire's. If the playback device's clock is a bit quicker (up to twice the speed) than the 6fire's one, the 6fire simply captures a frame (when neccesary) that contains two spdif frames. My question is (concerning capturing): How should I handle: - if 6fire's clock and playback device's clock differ, should I use playback device's clock (no interpolation of samples but capturing tool might have invalid sample rate information) or the 6fire's clock (samples need to be interpolated somehow)? - if the 6fire's clock should be used, are small clock drifts acceptable without interpolation? - in case of interpolation: how? - should I receive a whole spdif block (usually 192 spdif frames) and use the information I got from the block (f.ex. left/right channel bits) before copying it to alsa or should I directly copy every frame to alsa? Concerning (also output): Am I correct in assuming that SNDRV_PCM_FORMAT_IEC958_SUBFRAME_XX samples are formatted like in the specification (4 bit preamble, 24 bit audio data, VUCP flags)? If you know of any tool that is able to handle this format, please let me know so that I have a basis to test the whole stuff. Thanks, Torsten

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[alsa-devel] [PATCH] ASoC: soc-core: Ensure codec_reg has fixed length fields
by Dimitris Papastamos 01 Feb '11

01 Feb '11

Make the format of the codec_reg file more easily parsable. Remove the header field which gives the codec name. These changes are important when it comes to extend the debugfs codec_reg file to dump more than PAGE_SIZE bytes to make it easier to calculate offsets within the file. We still need to handle the case when the snd_soc_read() call fails and <no data: %d> is outputted. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 27 ++++++++++++++++++++++++--- 1 files changed, 24 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3feddd9..205cbd7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -69,10 +69,32 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +/* returns the minimum number of bytes needed to represent + * a particular given value */ +static int min_bytes_needed(unsigned long val) +{ + int c = 0; + int i; + + for (i = (sizeof val * 8) - 1; i >= 0; --i, ++c) + if (val & (1UL << i)) + break; + c = (sizeof val * 8) - c; + if (!c || (c % 8)) + c = (c + 8) / 8; + else + c /= 8; + return c; +} + /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { int ret, i, step = 1, count = 0; + int wordsize, regsize; + + wordsize = codec->driver->reg_word_size * 2; + regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; if (!codec->driver->reg_cache_size) return 0; @@ -80,12 +102,11 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (codec->driver->reg_cache_step) step = codec->driver->reg_cache_step; - count += sprintf(buf, "%s registers\n", codec->name); for (i = 0; i < codec->driver->reg_cache_size; i += step) { if (codec->readable_register && !codec->readable_register(codec, i)) continue; - count += sprintf(buf + count, "%2x: ", i); + count += sprintf(buf + count, "%.*x: ", regsize, i); if (count >= PAGE_SIZE - 1) break; @@ -101,7 +122,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (ret >= 0) count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", ret); + "%.*x", wordsize, ret); else count += snprintf(buf + count, PAGE_SIZE - count, -- 1.7.3.5

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[alsa-devel] [PATCH] ASoC: CX20442: fix NULL pointer dereference
by Janusz Krzysztofik 01 Feb '11

01 Feb '11

The CX20442 codec driver never provided the snd_soc_codec_driver's .reg_cache_default member. With the latest ASoC framework changes, it seems to be referred unconditionally, resulting in a NULL pointer dereference if missing. Provide it. Created and tested on Amstrad Delta against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- NOTE: I'm not sure if the value choosen is Amstrad Delta specific or not. sound/soc/codecs/cx20442.c | 3 +++ 1 file changed, 3 insertions(+) --- git/sound/soc/codecs/cx20442.c.orig 2011-01-31 20:09:18.000000000 +0100 +++ git/sound/soc/codecs/cx20442.c 2011-01-31 20:16:20.000000000 +0100 @@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct s return 0; } +static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC; + static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, + .reg_cache_default = &cx20442_reg, .reg_cache_size = 1, .reg_word_size = sizeof(u8), .read = cx20442_read_reg_cache,

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[alsa-devel] [PATCH] ASoC: Use snd_pcm_format_width() in snd_soc_params_to_frame_size()
by Mark Brown 01 Feb '11

01 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-utils.c | 23 +++-------------------- 1 files changed, 3 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 1d07b93..3f45e6a 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -28,26 +28,9 @@ int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) { int sample_size; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - case SNDRV_PCM_FORMAT_S16_BE: - sample_size = 16; - break; - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S20_3BE: - sample_size = 20; - break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S24_BE: - sample_size = 24; - break; - case SNDRV_PCM_FORMAT_S32_LE: - case SNDRV_PCM_FORMAT_S32_BE: - sample_size = 32; - break; - default: - return -ENOTSUPP; - } + sample_size = snd_pcm_format_width(params_format(params)); + if (sample_size < 0) + return sample_size; return snd_soc_calc_frame_size(sample_size, params_channels(params), 1); -- 1.7.2.3

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[alsa-devel] [PATCH] ARM: tegra: Add to struct harmony_audio_platform_data
by Stephen Warren 01 Feb '11

01 Feb '11

Add fields to describe all the audio-related GPIOs on Harmony, except for the codec's GPIO IRQ, which will be passed in its i2c_board_info. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, can you please apply this to your tegra-arch branch, and merge that. Thanks! arch/arm/mach-tegra/include/mach/harmony_audio.h | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-tegra/include/mach/harmony_audio.h b/arch/arm/mach-tegra/include/mach/harmony_audio.h index 5c46391..af08650 100644 --- a/arch/arm/mach-tegra/include/mach/harmony_audio.h +++ b/arch/arm/mach-tegra/include/mach/harmony_audio.h @@ -16,4 +16,7 @@ struct harmony_audio_platform_data { int gpio_spkr_en; + int gpio_hp_det; + int gpio_int_mic_en; + int gpio_ext_mic_en; }; -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Amstrad Delta: fix const related build error
by Janusz Krzysztofik 01 Feb '11

01 Feb '11

The Amstrad Delta ASoC driver used to override the digital_mute() callback, expected to be not provided by the on-board CX20442 CODEC driver, with its own implementation. While this is still posssible when substituting the whole empty snd_soc_dai_driver.ops member (the CX20442 case), replacing snd_soc_dai_ops.digital_mute only is no longer correct after the snd_soc_dai_driver.ops member has been constified, and results in build error. Drop this actually not used code path in hope the CX20442 driver never provides its own snd_soc_dai_ops structure. Created and tested against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Please consider queuing this patch for the current rc cycle. sound/soc/omap/ams-delta.c | 2 -- 1 file changed, 2 deletions(-) --- git/sound/soc/omap/ams-delta.c.orig 2011-01-31 19:49:29.000000000 +0100 +++ git/sound/soc/omap/ams-delta.c 2011-01-31 23:24:45.000000000 +0100 @@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct /* Set up digital mute if not provided by the codec */ if (!codec_dai->driver->ops) { codec_dai->driver->ops = &ams_delta_dai_ops; - } else if (!codec_dai->driver->ops->digital_mute) { - codec_dai->driver->ops->digital_mute = ams_delta_digital_mute; } else { ams_delta_ops.startup = ams_delta_startup; ams_delta_ops.shutdown = ams_delta_shutdown;

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[alsa-devel] [PATCH] ASoC: Tegra: Harmony: Remove redundant !!
by Stephen Warren 01 Feb '11

01 Feb '11

gpio_set_value* should accept logic values not just 0 or 1. The WM8903 GPIO driver has been fixed to work this way, so remove the redundant !! previously required when it didn't accept values >1. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/harmony.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index d1faa63..11e2cb8 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -132,7 +132,7 @@ static int harmony_event_int_spk(struct snd_soc_dapm_widget *w, struct harmony_audio_platform_data *pdata = harmony->pdata; gpio_set_value_cansleep(pdata->gpio_spkr_en, - !!SND_SOC_DAPM_EVENT_ON(event)); + SND_SOC_DAPM_EVENT_ON(event)); return 0; } -- 1.7.1

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[alsa-devel] [PATCH 1/2] ASoC: omap: rx51: Add stereo output support to audio jack
by Jarkko Nikula 01 Feb '11

01 Feb '11

Audio jack in Nokia RX-51/N900 is driven by TPA6130 headphone amplifier. This patch adds support for it and stereo output can be active when "Jack Function" == "TV-OUT" || "Headphone". As the TPA6130 can output very high volume levels the output is limited with snd_soc_limit_volume. Limiting value is found from Maemo kernel sources. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Just repost with Mark's ack added. Previous thread here: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-January/035312.ht… --- sound/soc/omap/Kconfig | 1 + sound/soc/omap/rx51.c | 43 +++++++++++++++++++++++++++++++++++++++---- 2 files changed, 40 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a088db6..b592298 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -24,6 +24,7 @@ config SND_OMAP_SOC_RX51 select OMAP_MCBSP select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X + select SND_SOC_TPA6130A2 help Say Y if you want to add support for SoC audio on Nokia RX-51 hardware. This is also known as Nokia N900 product. diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 09fb0df..251afbe 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -31,6 +31,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <plat/mcbsp.h> +#include "../codecs/tpa6130a2.h" #include <asm/mach-types.h> @@ -47,7 +48,8 @@ enum { RX51_JACK_DISABLED, - RX51_JACK_TVOUT, /* tv-out */ + RX51_JACK_TVOUT, /* tv-out with stereo output */ + RX51_JACK_HP, /* headphone: stereo output, no mic */ }; static int rx51_spk_func; @@ -57,6 +59,15 @@ static int rx51_jack_func; static void rx51_ext_control(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; + int hp = 0, tvout = 0; + + switch (rx51_jack_func) { + case RX51_JACK_TVOUT: + tvout = 1; + case RX51_JACK_HP: + hp = 1; + break; + } if (rx51_spk_func) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); @@ -66,9 +77,12 @@ static void rx51_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(dapm, "DMic"); else snd_soc_dapm_disable_pin(dapm, "DMic"); + if (hp) + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - gpio_set_value(RX51_TVOUT_SEL_GPIO, - rx51_jack_func == RX51_JACK_TVOUT); + gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout); snd_soc_dapm_sync(dapm); } @@ -153,6 +167,19 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w, return 0; } +static int rx51_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->dapm->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + tpa6130a2_stereo_enable(codec, 1); + else + tpa6130a2_stereo_enable(codec, 0); + + return 0; +} + static int rx51_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -212,11 +239,14 @@ static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = { static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), SND_SOC_DAPM_MIC("DMic", NULL), + SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event), }; static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HPLOUT"}, {"Ext Spk", NULL, "HPROUT"}, + {"Headphone Jack", NULL, "LLOUT"}, + {"Headphone Jack", NULL, "RLOUT"}, {"DMic Rate 64", NULL, "Mic Bias 2V"}, {"Mic Bias 2V", NULL, "DMic"}, @@ -224,7 +254,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static const char *spk_function[] = {"Off", "On"}; static const char *input_function[] = {"ADC", "Digital Mic"}; -static const char *jack_function[] = {"Off", "TV-OUT"}; +static const char *jack_function[] = {"Off", "TV-OUT", "Headphone"}; static const struct soc_enum rx51_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), @@ -265,6 +295,11 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) /* Set up RX-51 specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + err = tpa6130a2_add_controls(codec); + if (err < 0) + return err; + snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + snd_soc_dapm_sync(dapm); /* AV jack detection */ -- 1.7.2.3

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