November 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 01/32] ASoC: wm8350: Cleanup manual bias level transitions
by Lars-Peter Clausen 24 Nov '14

24 Nov '14

Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8350.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 628ec77..87f664b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1242,19 +1242,6 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8350_suspend(struct snd_soc_codec *codec) -{ - wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8350_resume(struct snd_soc_codec *codec) -{ - wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static void wm8350_hp_work(struct wm8350_data *priv, struct wm8350_jack_data *jack, u16 mask) @@ -1565,9 +1552,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, wm8350_mic_handler, 0, "Microphone detect", priv); - - wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } @@ -1596,8 +1580,6 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) * wait for its completion */ flush_delayed_work(&codec->dapm.delayed_work); - wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); - wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); return 0; @@ -1613,10 +1595,9 @@ static struct regmap *wm8350_get_regmap(struct device *dev) static struct snd_soc_codec_driver soc_codec_dev_wm8350 = { .probe = wm8350_codec_probe, .remove = wm8350_codec_remove, - .suspend = wm8350_suspend, - .resume = wm8350_resume, .get_regmap = wm8350_get_regmap, .set_bias_level = wm8350_set_bias_level, + .suspend_bias_off = true, .controls = wm8350_snd_controls, .num_controls = ARRAY_SIZE(wm8350_snd_controls), -- 1.8.0

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[alsa-devel] [PATCH 1/3] ASoC: alc5623: Cleanup bias level transitions
by Lars-Peter Clausen 24 Nov '14

24 Nov '14

Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Also remove the manual sequencing back to SND_SOC_BIAS_ON in resume as this is already handled by the ASoC core. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/alc5623.c | 22 +--------------------- 1 file changed, 1 insertion(+), 21 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 9d0755a..bdf8c5a 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -866,7 +866,6 @@ static int alc5623_suspend(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(alc5623->regmap, true); return 0; @@ -887,15 +886,6 @@ static int alc5623_resume(struct snd_soc_codec *codec) return ret; } - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* charge alc5623 caps */ - if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->dapm.bias_level = SND_SOC_BIAS_ON; - alc5623_set_bias_level(codec, codec->dapm.bias_level); - } - return 0; } @@ -906,9 +896,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) alc5623_reset(codec); - /* power on device */ - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (alc5623->add_ctrl) { snd_soc_write(codec, ALC5623_ADD_CTRL_REG, alc5623->add_ctrl); @@ -964,19 +951,12 @@ static int alc5623_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int alc5623_remove(struct snd_soc_codec *codec) -{ - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .probe = alc5623_probe, - .remove = alc5623_remove, .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, + .suspend_bias_off = true, }; static const struct regmap_config alc5623_regmap = { -- 1.8.0

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[alsa-devel] [PATCH 1/3] ASoC: uda134x: Remove is_powered_on_standby from platform data
by Lars-Peter Clausen 24 Nov '14

24 Nov '14

According to its documentation the is_powered_on_standby field of the uda134x platform data is supposed to prevent the the driver from shutting down the ADC and DAC in standby mode. This behavior was broken in commit commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support") almost 5 years ago and all the flag does now is cause the driver to go to SND_SOC_BIAS_ON in probe, just for the ASoC core to put it back into SND_SOC_BIAS_STANDBY right after probe. Apparently the intended behavior has not been missed, so just remove is_powered_on_standby from the platform data struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/uda134x.h | 12 ------------ sound/soc/codecs/uda134x.c | 5 +---- 2 files changed, 1 insertion(+), 16 deletions(-) diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index e475659..509efb0 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -18,18 +18,6 @@ struct uda134x_platform_data { struct l3_pins l3; void (*power) (int); int model; - /* - ALSA SOC usually puts the device in standby mode when it's not used - for sometime. If you unset is_powered_on_standby the driver will - turn off the ADC/DAC when this callback is invoked and turn it back - on when needed. Unfortunately this will result in a very light bump - (it can be audible only with good earphones). If this bothers you - set is_powered_on_standby, you will have slightly higher power - consumption. Please note that sending the L3 command for ADC is - enough to make the bump, so it doesn't make difference if you - completely take off power from the codec. - */ - int is_powered_on_standby; #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 32b2f78..54240f1 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -518,10 +518,7 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) uda134x_reset(codec); - if (pd->is_powered_on_standby) - uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); - else - uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (pd->model == UDA134X_UDA1341) { widgets = uda1341_dapm_widgets; -- 1.8.0

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[alsa-devel] [PATCH 1/3] ASoC: max98090: Fix digital microphone
by Jarkko Nikula 24 Nov '14

24 Nov '14

Commit e409dfbfccf9 ("ASoC: dapm: Add a few supply widget sanity checks") broke digital microphone support in max98090.c: max98090 i2c-193C9890:00: Conditional paths are not supported for supply widgets (DMICL_ENA -> [DMIC] -> DMIC Mux) max98090 i2c-193C9890:00: ASoC: no dapm match for DMICL_ENA --> DMIC --> DMIC Mux max98090 i2c-193C9890:00: ASoC: Failed to add route DMICL_ENA -> DMIC -> DMIC Mux max98090 i2c-193C9890:00: Conditional paths are not supported for supply widgets (DMICR_ENA -> [DMIC] -> DMIC Mux) max98090 i2c-193C9890:00: ASoC: no dapm match for DMICR_ENA --> DMIC --> DMIC Mux max98090 i2c-193C9890:00: ASoC: Failed to add route DMICR_ENA -> DMIC -> DMIC Mux Problem is partially caused by commit f69e3caa9e18 ("ASoC: max98090: Enable both DMIC channels also when using mono configuration") which connects "DMICL_ENA" and "DMICR_ENA" supply widgets to "DMIC Mux". Fix the breakage by reverting f69e3caa9e18 and then by adding additional "DMICR_ENA" to "DMICL" and "DMICL_ENA" to "DMICR" cross-connections. This disconnects these supply widgets from the mux and makes sure that both DMIC data channels are still enabled together. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max98090.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 2ad381c4ec57..19258a910974 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1311,6 +1311,10 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, + {"DMICL", NULL, "DMICL_ENA"}, + {"DMICL", NULL, "DMICR_ENA"}, + {"DMICR", NULL, "DMICL_ENA"}, + {"DMICR", NULL, "DMICR_ENA"}, {"DMICL", NULL, "AHPF"}, {"DMICR", NULL, "AHPF"}, @@ -1368,8 +1372,6 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"DMIC Mux", "ADC", "ADCR"}, {"DMIC Mux", "DMIC", "DMICL"}, {"DMIC Mux", "DMIC", "DMICR"}, - {"DMIC Mux", "DMIC", "DMICL_ENA"}, - {"DMIC Mux", "DMIC", "DMICR_ENA"}, {"LBENL Mux", "Normal", "DMIC Mux"}, {"LBENL Mux", "Loopback", "LTENL Mux"}, -- 2.1.3

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[alsa-devel] [PATCH] ASoC: rt5677: Fix the issue that the regmap_range "rt5677_ranges" cannot be accessed
by Oder Chiou 24 Nov '14

24 Nov '14

After the commit "19ba484d7b15c8650b30377aad6e65b34d3cf3d5", it will cause that the regmap_range "rt5677_ranges" cannot be accessed. The patch fixes this issue. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 37 ++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5677.h | 2 +- 2 files changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f2211f1..133010d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4287,6 +4287,7 @@ static int rt5677_probe(struct snd_soc_codec *codec) } mutex_init(&rt5677->dsp_cmd_lock); + mutex_init(&rt5677->dsp_pri_lock); return 0; } @@ -4344,10 +4345,19 @@ static int rt5677_read(void *context, unsigned int reg, unsigned int *val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_read(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_read(rt5677, RT5677_PRIV_DATA, val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + } + } else { regmap_read(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4357,10 +4367,20 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_write(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_DATA, + val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + } + } else { regmap_write(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4495,10 +4515,13 @@ static const struct regmap_config rt5677_regmap_physical = { .reg_bits = 8, .val_bits = 16, - .max_register = RT5677_VENDOR_ID2 + 1, + .max_register = RT5677_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5677_ranges) * + RT5677_PR_SPACING), .readable_reg = rt5677_readable_register, .cache_type = REGCACHE_NONE, + .ranges = rt5677_ranges, + .num_ranges = ARRAY_SIZE(rt5677_ranges), }; static const struct regmap_config rt5677_regmap = { diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index a02f64c..dbd9ffd 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1670,7 +1670,7 @@ struct rt5677_priv { struct rt5677_platform_data pdata; struct regmap *regmap, *regmap_physical; const struct firmware *fw1, *fw2; - struct mutex dsp_cmd_lock; + struct mutex dsp_cmd_lock, dsp_pri_lock; int sysclk; int sysclk_src; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: Arndale: Add machine device tree binding documentation for Arndale board
by Krishna Mohan Dani 24 Nov '14

24 Nov '14

Document the device tree binding for the Arndale board based machine driver. Signed-off-by: Krishna Mohan Dani <krishna.md(a)samsung.com> --- .../devicetree/bindings/sound/arndale.txt | 24 ++++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/arndale.txt diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt new file mode 100644 index 0000000..0e76946 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/arndale.txt @@ -0,0 +1,24 @@ +Audio Binding for Arndale boards + +Required properties: +- compatible : Can be the following, + "samsung,arndale-rt5631" + +- samsung,audio-cpu: The phandle of the Samsung I2S controller +- samsung,audio-codec: The phandle of the audio codec + +Optional: +- samsung,model: The name of the sound-card + +Arndale Boards has many audio daughter cards, one of them is +rt5631/alc5631. Below example shows audio bindings for rt5631/ +alc5631 based codec. + +Example: + +sound { + compatible = "samsung,arndale-rt5631"; + + samsung,audio-cpu = <&i2s0> + samsung,audio-codec = <&rt5631>; +}; -- 1.7.9.5

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[alsa-devel] TLV320AIC3104 Sample rate problem
by Catalin Crenguta 24 Nov '14

24 Nov '14

Hello, I'm using an Audio Cape Rev. B connected to BeagleBone Black (MCLK=12MHz) and I'm trying to record audio at sample rate 8KHz. Recording and playback using ALSA is OK, but when playing the recorded samples on PC you can hear that the actual sample rate is lower (the pitch shifted to a lower tone). I have added debug messages throughout the alsa driver tlv320aic3x.c, and I found out that the PLL parameters are computed correctly as in TLV320AIC3104 Datasheet Table found at page 27. (p: 1 j: 8 d: 1920 r: 1). Has anyone else encountered a problem like this? Can someone with more experience check the driver? Kernel: https://github.com/beagleboard/linux , commit 2bcc89c, 3.14.19+. Best regards, Catalin Crenguta

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[alsa-devel] [PATCH 00/29 v2] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 24 Nov '14

24 Nov '14

This patchset is to enable ALSA Dice driver to handle more models, to improve ALSA firewire-speakers driver as OXFW driver. This patchset is for linux-next branch and all of my work for Linux 3.19. == Features: Currently, ALSA Dice driver is limited for playback-only devices. The aim of this patchset is to add supports for: - Full-duplex streams with synchronization - Both PCM capture and playback - Both MIDI capture and playback Currently, ALSA firewire-speakers driver is only for two playback-only models. And this driver has a bug for one of them. The aim of this patchset is to fix the bug and to add supports for more models with: - Both PCM capture and playback - Both MIDI capture and playback The renew drivers doesn't support: - Synchronization between multiple devices on the same bus For this purpose, please join in user-space driver project (FFADO) - Any control functionality for internal DSP For this purpose, please use ffado-dbus-server/ffado-mixer, or write your own application == Updates from my previous post: [alsa-devel] [PATCH 00/14] ALSA: Dice enhancement http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/081890.ht… - Fix typo - Minor code change [alsa-devel] [RFC][PATCH 00/15 v4] OXFW driver, a succesor to firewire-speakers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-May/076581.html - Support more Mackie (Loud) models by name detection - Fix a bug to handle FireWave - Use AV/C Stream Format Information command to set sampling rate == Remarks: - When the Dice driver is loaded, user-space driver fails to register notification address, thus cannot handle device for streaming. See: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081731.… - OXFW chipset seems to transmit wrong values for timestamp, thus the driver don't apply full-duplex streams with synchronization == Supported Dice-based models (as long as I know): TC Electronic - Desktop Konnekt 6 - Impact Twin - Konnekt 8/24D/Live - Studio Konnekt 48 PreSonus - FireStudio 26x26/Mobile/Project/Tube/LightPipe - StudioLive 1602/1642/2442 - FireStation Mackie - Onyx Blackbird - Onyx i-series (latter models) Focusrite - Saffire Pro 14/24/24 DSP/26/40/56 M-Audio - Profire 610/2626 Alessis - IO 24/26 - MultiMix 12/16 Weiss - ADC2 - Vesta - Mineva - AFI 1 - TAG DAC 1 - INT 202 - DAC 202 - Flex 5000 DnR - Axus FireWire I/O card 16x16 Lexicon - I-ONIX FW 810S Allen and Heath - Zed R16 Midas - Venice F32 == Supported OXFW-based models (as long as I know): Griffin: - FireWave LaCie: - Firewire Speakers Behringer: - F-Control Audio 202 Mackie (Loud): - Onyx-i series (former models) - Onyx Satellite - Tapco Link.Firewire - d.2 pro/d.4 pro - U.420/U.420d Takashi Sakamoto (29): ALSA: dice: Rename structure and its members ALSA: dice: Move file to its own directory ALSA: dice: Split transaction functionality into a file ALSA: dice: Split stream functionality into a file ALSA: dice: Split PCM functionality into a file ALSA: dice: Split hwdep functionality into a file ALSA: dice: Split proc interface into a file ALSA: dice: Add new functions for constraints of PCM parameters ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: dice: remove experimental state ALSA: speakers: Rename to oxfw and rename some members ALSA: oxfw: Move to its own directory ALSA: oxfw: Split stream functionality to a new file and add a header file ALSA: oxfw: Split PCM functionality to a new file ALSA: oxfw: Split control functionality to a new file ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 26 +- sound/firewire/Makefile | 6 +- sound/firewire/dice.c | 1511 ---------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-hwdep.c | 190 ++++ sound/firewire/{ => dice}/dice-interface.h | 0 sound/firewire/dice/dice-midi.c | 176 ++++ sound/firewire/dice/dice-pcm.c | 404 ++++++++ sound/firewire/dice/dice-proc.c | 252 +++++ sound/firewire/dice/dice-stream.c | 423 ++++++++ sound/firewire/dice/dice-transaction.c | 382 +++++++ sound/firewire/dice/dice.c | 359 +++++++ sound/firewire/dice/dice.h | 189 ++++ sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw-command.c | 153 +++ sound/firewire/oxfw/oxfw-control.c | 283 ++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++ sound/firewire/oxfw/oxfw-midi.c | 189 ++++ sound/firewire/oxfw/oxfw-pcm.c | 399 ++++++++ sound/firewire/oxfw/oxfw-proc.c | 113 +++ sound/firewire/oxfw/oxfw-stream.c | 702 +++++++++++++ sound/firewire/oxfw/oxfw.c | 310 ++++++ sound/firewire/oxfw/oxfw.h | 145 +++ sound/firewire/speakers.c | 792 --------------- 26 files changed, 4887 insertions(+), 2319 deletions(-) delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-hwdep.c rename sound/firewire/{ => dice}/dice-interface.h (100%) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/dice/dice-pcm.c create mode 100644 sound/firewire/dice/dice-proc.c create mode 100644 sound/firewire/dice/dice-stream.c create mode 100644 sound/firewire/dice/dice-transaction.c create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw-command.c create mode 100644 sound/firewire/oxfw/oxfw-control.c create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c create mode 100644 sound/firewire/oxfw/oxfw-midi.c create mode 100644 sound/firewire/oxfw/oxfw-pcm.c create mode 100644 sound/firewire/oxfw/oxfw-proc.c create mode 100644 sound/firewire/oxfw/oxfw-stream.c create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h delete mode 100644 sound/firewire/speakers.c -- 1.9.1

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[alsa-devel] [PATCH] ASoC: bt-sco: support wide band speech
by Barry Song 24 Nov '14

24 Nov '14

From: Yibo Cai <yibo.cai(a)csr.com> HFP1.6 requires support for wide band speech(16KHz) over Bluetooth. Signed-off-by: Yibo Cai <yibo.cai(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/bt-sco.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index c4cf069..fc0664b 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -31,14 +31,14 @@ static struct snd_soc_dai_driver bt_sco_dai = { .stream_name = "Playback", .channels_min = 1, .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; -- 2.1.1

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[alsa-devel] [PATCH] ASoC: rt5631: Fixing compilation warning when DT is disabled
by Krishna Mohan Dani 24 Nov '14

24 Nov '14

Fixes the following compilation warning: Warning: 'rt5631_i2c_dt_ids' defined but not used - when DT is not used. Signed-off-by: Claude Youn <claude.youn(a)gmail.com> Signed-off-by: Krishna Mohan Dani <krishna.md(a)samsung.com> --- sound/soc/codecs/rt5631.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 3b7d5e4..9425545 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1691,12 +1691,14 @@ static const struct i2c_device_id rt5631_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); +#ifdef CONFIG_OF static struct of_device_id rt5631_i2c_dt_ids[] = { { .compatible = "realtek,rt5631"}, { .compatible = "realtek,alc5631"}, { } }; MODULE_DEVICE_TABLE(of, rt5631_i2c_dt_ids); +#endif static const struct regmap_config rt5631_regmap_config = { .reg_bits = 8, -- 1.7.9.5

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