June 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC][PATCH 00/11] improve support for DM1500/BeBoBv3
by Takashi Sakamoto 08 Jun '15

08 Jun '15

Hi, This patchset is to improve support for DM1500 and BeBoBv3; DM1500 is an ASIC which ArchWave produced. BeBoB is an interface protocol for DM series firmware. Currently, Some devices are reported not to work with current ALSA BeBoB driver. [FFADO-user] Support Behringer UFX1604 http://sourceforge.net/p/ffado/mailman/message/32846563/ [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels http://mailman.alsa-project.org/pipermail/alsa-devel/2015-April/090967.html They commonly install BeBoB version 3 firmware. As long as seeing these devices, they may have below quirks comparing to version 1 and 2. Furthermore, DM1500 seems to have its own quirks. The first half of this patchset is for improvement of clock detection, the rest is for quirks. The latter is tested by Kim Tore Jensen (Thanks!), while the first half is not widely tested. I hope someone to test it till opening next merge window (but just one week rest!). I push these patches into my own repository. It's DKMS available. https://github.com/takaswie/snd-firewire-improve/tree/bebob3 Regards Takashi Sakamoto (11): ALSA: bebob: improve signal mode detection for clock source ALSA: bebob: apply new enumerator to normalize the type of clock source ALSA: bebob: preparation for replacing string literals by normalized representation for model-dependent structures ALSA: bebob: use normalized representation for the type of clock source ALSA: bebob: obsolete string literal expression for clock source ALSA: bebob: add SYT-Match support ALSA: bebob: add 'version' member for BeBoB protocol version ALSA: bebob: expand timeout for DM1500 quirk ALSA: bebob: loosen up severity of checking continuity for BeBoB v3 quirk ALSA: bebob: keep duplex streams always to keep internal multiplexer properly ALSA: bebob: add support for Behringer FCA 610/1616 sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 14 ++- sound/firewire/bebob/bebob.h | 17 ++-- sound/firewire/bebob/bebob_focusrite.c | 33 +++++--- sound/firewire/bebob/bebob_maudio.c | 23 +++-- sound/firewire/bebob/bebob_midi.c | 8 +- sound/firewire/bebob/bebob_pcm.c | 14 +-- sound/firewire/bebob/bebob_proc.c | 22 ++--- sound/firewire/bebob/bebob_stream.c | 150 ++++++++++++++++++++++++--------- sound/firewire/bebob/bebob_terratec.c | 30 +++++-- sound/firewire/bebob/bebob_yamaha.c | 20 ++++- 11 files changed, 233 insertions(+), 99 deletions(-) -- 2.1.4

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[alsa-devel] Intel HDA & Realtek issues on ASUS G751JM
by Nicolae Rosia 08 Jun '15

08 Jun '15

Hello, I can't get 4.0/5.1 sound working on my laptop [0] [1] with Fedora 22. With Linux 4.0 there is no output on jacks but with 4.1-rc6 I'm getting sound only on one output. Switching between pacmd set-card-profile alsa_card.pci-0000_00_1b.0 output:analog-stereo and pacmd set-card-profile alsa_card.pci-0000_00_1b.0 output:analog-surround-40 only boosts the volume on the first output. Attached is dmesg, lsmod, result of lspci -vv and pacmd list-cards and list-sinks and output of various information from /proc/asound while runnig 4.1-rc6. Let me know if you need more information or how I can help. Best regards, Nicolae Rosia [rosia@localhost ~]$ cat /proc/asound/devices 1: : sequencer 2: [ 0] : control 3: [ 0- 3]: digital audio playback 4: [ 0- 7]: digital audio playback 5: [ 0- 8]: digital audio playback 6: [ 0- 0]: hardware dependent 7: [ 1] : control 8: [ 1- 0]: digital audio playback 9: [ 1- 0]: digital audio capture 10: [ 1- 1]: digital audio playback 11: [ 1- 0]: hardware dependent 33: : timer [rosia@localhost ~]$ cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xeda14000 irq 33 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xeda10000 irq 32 [rosia@localhost ~]$ cat /proc/asound/modules 0 snd_hda_intel 1 snd_hda_intel [rosia@localhost ~]$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k4.1.0-0.rc6.git2.1.fc23.x86_64. [0] http://www.emag.ro/laptop-gaming-asus-cu-procesor-intelr-coretm-i7-4710hq-2… [1] https://www.asus.com/Notebooks_Ultrabooks/ASUS_ROG_G751JM/specifications/

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[alsa-devel] [PATCH v2 00/23] ASoC: tas2552: Fixes, cleanups and improvements
by Peter Ujfalusi 08 Jun '15

08 Jun '15

Hi, Changes since v01: - missing patch found. It was merged with the previous patch while fixing checkpatch errors (patch 19 got squashed to patch 18) - Sorry for this. I have used the TAS2552-EVM to verify that my setup is in working condition for a work on other audio product. During the bringup of the EVM I have faced some issues and limitation with the driver. The result is this series against the tas2552 driver. Notable changes: - Fixes for kernel crashes - The enable-gpio was not optional as it was documented in the DTS - Register bit definitions were not correct in several places - Adding recommended startup/stop sequence - Handling of sample formats the driver claims to support - PLL configuration fixes and reference clock selection - TDM support - Gain control fix and control to select DIN source Regards, Peter --- Peter Ujfalusi (23): ASoC: tas2552: Make the enable-gpio really optional ASoC: tas2552: Fix kernel crash when the codec is loaded but not part of a card ASoC: tas2552: Fix kernel crash caused by wrong kcontrol entry ASoC: tas2552: Correct PDM configuration register bit definitions ASoC: tas2552: Correct CFG1 register bit definitions ASoC: tas2552: Simplify the tas2552_mute function ASoC: tas2552: Simplify and reverse the functionality of tas2552_sw_shutdown ASoC: tas2552: Rename mclk parameter to pll_clkin to match with the datasheet DT/sound: bindings header file for tas2552 codec ASoC: tas2552: Add support for pll and pdm source clock selection ASoC: tas2552: Correct dai format support ASoC: tas2552: Correct and clean up data format and BCLK/WCLK direction ASoC: tas2552: Add TDM support ASoC: tas2552: Clean up the Digital - Analog DAPM route definition ASoC: tas2552: Correct the Speaker Driver Playback Volume (PGA_GAIN) ASoC: tas2552: Implement startup/stop sequence as per TRM ASoC: tas2552: Add support for word length configuration ASoC: tas2552: Configure the WCLK frequency based on the stream ASoC: tas2552: Correct the PLL configuration ASoC: tas2552: Add control for selecting DIN source ASoC: tas2552: Correct Output Data register usage ASoC: tas2552: Correct Boost Auto-Pass Through Control register usage ASoC: tas2552: Code, define alignment changes for uniformity include/dt-bindings/sound/tas2552.h | 18 ++ sound/soc/codecs/tas2552.c | 435 +++++++++++++++++++++++++++--------- sound/soc/codecs/tas2552.h | 149 ++++++------ 3 files changed, 431 insertions(+), 171 deletions(-) create mode 100644 include/dt-bindings/sound/tas2552.h -- 2.4.2

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[alsa-devel] dshare plugin issue in alsa-lib.1.0.27
by Srinivasan S 08 Jun '15

08 Jun '15

Dear Clemens, Thanks a lot for all your suggestions w.r.t the thread [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS Could you please provide some suggestions or guidance w.r.t the below issue Now am using linux kernel version 3.14 with alsa-lib.1.0.27 I feel dshare plugin is not working now with this linux kernel & alsa-lib version , When I try to run theaplay -D VOUTR NeroSoundTrax_test8_PCM_Mono_VBR_16SS_48000Hz.wav I don't get any errors, but it seems that audio is playing but I don't hear any audio But when I run the same in linux kernel version 3.12 with alsa-lib.1.0.26, I hear the audio output wav @am335x-evm:/# aplay -D VOUTR NeroSoundTrax_test8_PCM_Mono_VBR_16SS_48000Hz.w [ 876.513092] DAVINCIIIIIIIIII UDA134XXXXX SYSCLK=12288000 [ 876.518771] DAVINCIIIIIIIIII UDA134XXXXX BCLK FREQQ=1536000 [ 876.524777] DAVINCIIIIIIIIII UDA134XXXXX SYSCLK/BCLK_FREQ =8 [ 876.530813] uda134x_hw_params CLOCKS uda134x_hw_params uda134x->sysclk: 12288000, params_rate(params):48000 [ 876.541191] uda134x_hw_params FORMATS uda134x_hw_params dai_fmt: 16385, params_format:2 [ 876.549720] uda134x_hw_params FORMATS uda134x_hw_params uda134x->sysclk / params_rate(params) 256 [ 876.559165] UDA1345TSSSSSSSSSSSS SYSCLK / fs ratio is 256 [ 876.564929] uda134x_hw_params dai_fmt: 16385, params_format:2 [ 876.571029] UDA1345TSSSSSSSSSSSS FORMAT SND_SOC_DAIFMT_I2S [ 876.577006] ENTERED davinci_config_channel_size davinci_config_channel_size: tx_rotate = 4 [ 876.585812] ENTERED davinci_config_channel_size davinci_config_channel_size: MASK= 65535 [ 876.598147] uda134x_unnnnnnnnmuteeeeeeeeee uda134x_mute mute: 0 [ 876.605173] davinci_mcasp_starttttttttttttttttttttt SNDRV_PCM_STREAM_PLAYBACK Playing WAVE 'NeroSoundTrax_test8_PCM_Mono_VBR_16SS_48000Hz.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono I have tired the following: Firstly, I have tried with basic arecord & aplay in linux kernel version 3.14 that works, without the above asound.conf I have even tried with combination's of linux kernel version 3.14 & alsa-lib-1.0.26 & linux kernel verison 3.12 with alsa-lib-1.0.27 that didn't work, with the above asound.conf Now I feel, it seems to be there is an dependency issue with alsa-lib.1.0.27 So, could you please try from your end with linux kernel version 3.14 with the above asound.conf with above alsalib version & please let me know whether it works or not? If not, Could you please let me know how to get this thing working with linux kernel 3.14 (as all my development activities are being done in linux 3.14) Kindly do the needful as early as possible, as am stuck with this from many day's Many Many Thanks in advance,

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[alsa-devel] [PATCH] usb-audio: add native DSD support for JLsounds I2SoverUSB
by Jurgen Kramer 08 Jun '15

08 Jun '15

This patch adds native DSD support for the XMOS based JLsounds I2SoverUSB board Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2917534..04c285c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1266,8 +1266,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; - /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ - case USB_ID(0x20b1, 0x2009): + + case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ + case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */ if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.1.0

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[alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Łukasz Wojniłowicz 07 Jun '15

07 Jun '15

Hi all, My problem is that when I set channel output to "Analog output surround 5.1" (or 2.1 or 4.1), I hear no sound in front speakers, but in case of 5.1 or 4.1 I can hear sound in rear speakers. I use kernel 3.19.5 and it didn't happen in earlier kernels that is 3.18 and in 3.17 for sure. My configuration is: OS: Fedora 21 64 bit SoundCard: ALC888 Model name: acer-aspire-4930g Laptop: Acer Aspire 5930g Speakers: external 5.1 with 3 jacks Sound channel configuration module is standard configuration module from KDE. Additional info: 1) When channel output is 5.1 and I unplug green jack and leave two others jacks connected then I can hear sound in laptop internal speakers. 2) Earlier (kernel 3.18 and 3.17) I didn't have possibility to choose 2.1 configuration and additionally to 4.1 configuration I could also choose 4.0 configuration. Thanks for looking into problem. Cheers LW

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[alsa-devel] [PATCH v2 0/2] Update ALSA driver to use media controller API
by Shuah Khan 06 Jun '15

06 Jun '15

This patch series updates ALSA driver to use media controller API to share tuner with DVB and V4L2 drivers that control AU0828 media device. Two new interfaces are added to media controller API to enable creating media device as a device resource. This allows creating media device as a device resource on the main struct device that is common to all drivers that control a single media hardware and share resources on it. Drivers then can find the common media device to register entities and manage links, and pipelines. Tested with and without CONFIG_MEDIA_CONTROLLER enabled. Tested tuner entity doesn't exist case as au0828 v4l2 driver is the one that will create the tuner when it gets updated to use media controller API. Please note that au0828 updates media controller are necessary before the resource sharing can work across ALSA and au0828 dvb and v4l2 drivers. This work is in progress and another developer is working on it. Changes since v1: - Moved media specific code into new header and source file - Created a new structure for media controller specific fields - Added a new define USE_MEDIA_CONTROLLER to add media specific code. This new define is defined only when Media Controller support is enabled. New media specific code is active only when this define is defined. - Moved media_deviced_delete() check under !was_shutdown block - Removed return check for media_device_init() - Changed new Media Controller interface names for clarity. Now they are named media_device_get_devres() and media_device_find_devres(). - Calling media_entity_pipeline_stop() on a pipe that isn't active appears to be safe. This needs further testing once AU0828 media controller patches are ready. Shuah Khan (2): media: new media controller API for device resource support sound/usb: Update ALSA driver to use media controller API drivers/media/media-device.c | 33 ++++++++ include/media/media-device.h | 2 + sound/usb/Makefile | 3 +- sound/usb/card.c | 5 ++ sound/usb/card.h | 1 + sound/usb/media.c | 181 +++++++++++++++++++++++++++++++++++++++++++ sound/usb/media.h | 40 ++++++++++ sound/usb/pcm.c | 10 ++- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 ++- sound/usb/stream.c | 3 + sound/usb/usbaudio.h | 1 + 12 files changed, 286 insertions(+), 3 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

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[alsa-devel] [RFC PATCH] ASoC: wm8741: Allow master clock switching
by Sergej Sawazki 05 Jun '15

05 Jun '15

The set of supported sample rates depends on the master clock supplied to the codec. Allow the machine driver to set the required master clock in hw_params(). Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- This patch has been tested on a board with two master clocks, 22.5792MHz (for 44.1kHz, 88.2kHz, ...) and 24.576MHz (for 48kHz, 96kHz, ...). sound/soc/codecs/wm8741.c | 63 +++++++++++++---------------------------------- 1 file changed, 17 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 09ff01f..47c5577 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -125,18 +125,6 @@ static const struct snd_soc_dapm_route wm8741_dapm_routes[] = { { "VOUTRN", NULL, "DACR" }, }; -static struct { - int value; - int ratio; -} lrclk_ratios[WM8741_NUM_RATES] = { - { 1, 128 }, - { 2, 192 }, - { 3, 256 }, - { 4, 384 }, - { 5, 512 }, - { 6, 768 }, -}; - static const unsigned int rates_11289[] = { 44100, 88200, }; @@ -209,29 +197,6 @@ static const struct snd_pcm_hw_constraint_list constraints_36864 = { .list = rates_36864, }; - -static int wm8741_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); - - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC - enforce this. - */ - if (!wm8741->sysclk) { - dev_err(codec->dev, - "No MCLK configured, call set_sysclk() on init\n"); - return -EINVAL; - } - - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - wm8741->sysclk_constraints); - - return 0; -} - static int wm8741_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -241,17 +206,24 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; int i; - /* Find a supported LRCLK ratio */ - for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { - if (wm8741->sysclk / params_rate(params) == - lrclk_ratios[i].ratio) + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8741->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on hw_params\n"); + return -EINVAL; + } + + /* Find a supported LRCLK rate */ + for (i = 0; i < wm8741->sysclk_constraints->count; i++) { + if (wm8741->sysclk_constraints->list[i] == params_rate(params)) break; } - /* Should never happen, should be handled by constraints */ - if (i == ARRAY_SIZE(lrclk_ratios)) { - dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n", - wm8741->sysclk / params_rate(params)); + if (i == wm8741->sysclk_constraints->count) { + dev_err(codec->dev, "LRCLK %d unsupported with MCLK %d\n", + params_rate(params), wm8741->sysclk); return -EINVAL; } @@ -274,8 +246,8 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d", - params_width(params)); + dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d, rate param = %d", + params_width(params), params_rate(params)); snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); return 0; @@ -403,7 +375,6 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8741_dai_ops = { - .startup = wm8741_startup, .hw_params = wm8741_hw_params, .set_sysclk = wm8741_set_dai_sysclk, .set_fmt = wm8741_set_dai_fmt, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: wm8731: initialize the hardware when loading the codec driver
by Sergey Kiselev 05 Jun '15

05 Jun '15

From: Sergey Kiselev <sergey.kiselev(a)intel.com> This patch moves the requesting supplies, hardware reset and initialization from wm8731_probe to wm8731_i2c_probe and wm8731_spi_probe. So that the codec hardware is initialized when loading the codec driver, and not when loading the machine driver. This avoids unnecesary hardware resets and re-initializations when re-loading the machine driver. Signed-off-by: Sergey Kiselev <sergey.kiselev(a)intel.com> --- sound/soc/codecs/wm8731.c | 80 ++++++++++++++++++++++++++++++----------------- 1 file changed, 52 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 915ea11..28401f6 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -84,7 +84,7 @@ static bool wm8731_writeable(struct device *dev, unsigned int reg) return reg <= WM8731_RESET; } -#define wm8731_reset(c) snd_soc_write(c, WM8731_RESET, 0) +#define wm8731_reset(m) regmap_write(m, WM8731_RESET, 0) static const char *wm8731_input_select[] = {"Line In", "Mic"}; @@ -571,44 +571,54 @@ static struct snd_soc_dai_driver wm8731_dai = { .symmetric_rates = 1, }; -static int wm8731_probe(struct snd_soc_codec *codec) +static int wm8731_request_supplies(struct device *dev, + struct wm8731_priv *wm8731) { - struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret = 0, i; for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++) wm8731->supplies[i].supply = wm8731_supply_names[i]; - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8731->supplies), + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(wm8731->supplies), wm8731->supplies); if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + dev_err(dev, "Failed to request supplies: %d\n", ret); return ret; } ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + dev_err(dev, "Failed to enable supplies: %d\n", ret); return ret; } - ret = wm8731_reset(codec); + return 0; +} + +static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) +{ + int ret = 0; + + ret = wm8731_reset(wm8731->regmap); if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + dev_err(dev, "Failed to issue reset: %d\n", ret); goto err_regulator_enable; } - snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Clear POWEROFF, keep everything else disabled */ + regmap_write(wm8731->regmap, WM8731_PWR, 0x7f); /* Latch the update bits */ - snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0); - snd_soc_update_bits(codec, WM8731_ROUT1V, 0x100, 0); - snd_soc_update_bits(codec, WM8731_LINVOL, 0x100, 0); - snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_LOUT1V, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_ROUT1V, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_LINVOL, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_RINVOL, 0x100, 0); /* Disable bypass path by default */ - snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0); + regmap_update_bits(wm8731->regmap, WM8731_APANA, 0x8, 0); + + regcache_mark_dirty(wm8731->regmap); /* Regulators will have been enabled by bias management */ regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -621,19 +631,7 @@ err_regulator_enable: return ret; } -/* power down chip */ -static int wm8731_remove(struct snd_soc_codec *codec) -{ - struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { - .probe = wm8731_probe, - .remove = wm8731_remove, .set_bias_level = wm8731_set_bias_level, .suspend_bias_off = true, @@ -690,6 +688,12 @@ static int wm8731_spi_probe(struct spi_device *spi) mutex_init(&wm8731->lock); + spi_set_drvdata(spi, wm8731); + + ret = wm8731_request_supplies(&spi->dev, wm8731); + if (ret != 0) + return ret; + wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); @@ -698,7 +702,9 @@ static int wm8731_spi_probe(struct spi_device *spi) return ret; } - spi_set_drvdata(spi, wm8731); + ret = wm8731_hw_init(&spi->dev, wm8731); + if (ret != 0) + return ret; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8731, &wm8731_dai, 1); @@ -712,7 +718,12 @@ static int wm8731_spi_probe(struct spi_device *spi) static int wm8731_spi_remove(struct spi_device *spi) { + struct wm8731_priv *wm8731 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); + + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + return 0; } @@ -754,6 +765,12 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, mutex_init(&wm8731->lock); + i2c_set_clientdata(i2c, wm8731); + + ret = wm8731_request_supplies(&i2c->dev, wm8731); + if (ret != 0) + return ret; + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); @@ -762,7 +779,9 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, return ret; } - i2c_set_clientdata(i2c, wm8731); + ret = wm8731_hw_init(&i2c->dev, wm8731); + if (ret != 0) + return ret; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8731, &wm8731_dai, 1); @@ -776,7 +795,12 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, static int wm8731_i2c_remove(struct i2c_client *client) { + struct wm8731_priv *wm8731 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + return 0; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5645: Init jack_detect_work before registering irq
by Nicolas Boichat 05 Jun '15

05 Jun '15

Prevents frequent panic on boot, if the irq handler rt5645_irq gets called before the workqueue rt5645_jack_detect_work is initialized. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- Applies on top of broonie/sound.git/for-next (dac48b88824392de29a437607b7dbf844bb66f74) sound/soc/codecs/rt5645.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e9544c6..d5f0f56 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3416,6 +3416,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } } + INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); + if (rt5645->i2c->irq) { ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING @@ -3434,8 +3436,6 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Fail gpio_direction hp_det_gpio\n"); } - INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, rt5645_dai, ARRAY_SIZE(rt5645_dai)); } -- 2.2.0.rc0.207.ga3a616c

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