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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

2 1

[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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[alsa-devel] [PATCH] ASoC: rt5677: Avoid recursive locking in the DEADLOCK detector
by Oder Chiou 05 Jun '15

05 Jun '15

The patch is for to avoid recursive locking in the DEADLOCK detector. In the driver, it encountered the warnning message of the recursive locking in the function "regmap_lock_mutex". Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 31d969a..ee5b570 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4998,6 +4998,22 @@ static const struct regmap_config rt5677_regmap_physical = { .num_ranges = ARRAY_SIZE(rt5677_ranges), }; +static void rt5677_regmap_lock_mutex(void *__lock) +{ + struct mutex *lock = __lock; + + mutex_lock(lock); +} + +static void rt5677_regmap_unlock_mutex(void *__lock) +{ + struct mutex *lock = __lock; + + mutex_unlock(lock); +} + +static struct mutex rt5677_regmap_lock; + static const struct regmap_config rt5677_regmap = { .reg_bits = 8, .val_bits = 16, @@ -5009,6 +5025,9 @@ static const struct regmap_config rt5677_regmap = { .readable_reg = rt5677_readable_register, .reg_read = rt5677_read, .reg_write = rt5677_write, + .lock = rt5677_regmap_lock_mutex, + .unlock = rt5677_regmap_unlock_mutex, + .lock_arg = &rt5677_regmap_lock, .cache_type = REGCACHE_RBTREE, .reg_defaults = rt5677_reg, @@ -5206,6 +5225,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, return ret; } + mutex_init(&rt5677_regmap_lock); + regmap_read(rt5677->regmap, RT5677_VENDOR_ID2, &val); if (val != RT5677_DEVICE_ID) { dev_err(&i2c->dev, -- 1.9.1

4 5

[alsa-devel] [PATCH v2] ASoC: dapm: fix prefixed stream name matching in snd_soc_dapm_link_dai_widgets
by Arnaud Mouiche 05 Jun '15

05 Jun '15

commit "ASoC: dapm: Modify widget stream name according to prefix" fixed a part of the problem when dai_w->name and w->sname are equal before being prefixed. Remains the case where w->sname is only a sub string of dai_w->name. For example, TLV320AIC3x codec defines the widget SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_PWR, 7, 0) In snd_soc_dapm_link_dai_widgets, we will failed to match "[prefix] Left Playback" with "[prefix] Playback". Since dai_w and w necessarily belongs to the same dapm when strstr() is called, they necessarily have the same prefix which allow us to compare 'w->sname + prefix_len' with 'dai_w->name + prefix_len' directly without length check. Signed-off-by: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> --- sound/soc/soc-dapm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e19a676..fc4408b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3392,6 +3392,9 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { + const char *prefix = soc_dapm_prefix(dai_w->dapm); + size_t prefix_len = prefix ? strlen(prefix)+1 : 0; + switch (dai_w->id) { case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: @@ -3415,7 +3418,8 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) break; } - if (!w->sname || !strstr(w->sname, dai_w->name)) + if (!w->sname || + !strstr(w->sname + prefix_len, dai_w->name + prefix_len)) continue; if (dai_w->id == snd_soc_dapm_dai_in) { -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: dapm: fix prefixed stream name matching in snd_soc_dapm_link_dai_widgets
by Arnaud Mouiche 05 Jun '15

05 Jun '15

commit "ASoC: dapm: Modify widget stream name according to prefix" fixed a part of the problem when dai_w->name and w->sname are equal before being prefixed. Remains the case where w->sname is only a sub string of dai_w->name. For example, TLV320AIC3x codec defines the widget SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_PWR, 7, 0) In snd_soc_dapm_link_dai_widgets, we will failed to match "[prefix] Left Playback" with "[prefix] Playback". Since dai_w and w necessarily belongs to the same dapm when strstr() is called, they necessarily have the same prefix which allow us to compare 'w->sname + prefix_len' with 'dai_w->name + prefix_len' directly without length check. Signed-off-by: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> --- sound/soc/soc-dapm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e19a676..7194df4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3392,6 +3392,9 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { + const char *prefix = soc_dapm_prefix(dai_w->dapm); + size_t prefix_len = prefix ? strlen(prefix)+1 : 0; + switch (dai_w->id) { case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: @@ -3416,6 +3419,8 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) } if (!w->sname || !strstr(w->sname, dai_w->name)) + if (!w->sname || + !strstr(w->sname + prefix_len, dai_w->name + prefix_len)) continue; if (dai_w->id == snd_soc_dapm_dai_in) { -- 1.9.1

2 1

Re: [alsa-devel] (alsa-lib)pcm_hw.c: Invalid PCM state for chmap_get: OPEN
by Raymond Yau 05 Jun '15

05 Jun '15

>> >> Have you tested surround21? > > Yes, front speakers work fine, however LFE doesn't seem to work. Instead when I try to test LFE (from sound UI) I have weird noise in one of rear speakers. There are no such problems with any other mode though (I see that officially only 5.1 mode is supported, however 4.1, 5.0 and 5.1 all work fine), I decided to leave 2.1 mode for now, not sure whose fault is that. http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=3c4a22ea49a881cd… It is strange that pcm state is open ( 0.089| 0.001) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-21 ( 0.089| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 2.1 (analog-surround-21) ( 0.089| 0.000) D: [pulseaudio] alsa-util.c: Trying surround21:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.089| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: Invalid PCM state for chmap_get: OPEN ( 0.089| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround21:1 ( 0.089| 0.000) I: [pulseaudio] alsa-util.c: Disabling tsched mode since BATCH flag is set ( 0.089| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 1213 ms ( 0.090| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 4797 samples), period size second (to 1199 samples). ( 0.090| 0.000) I: [pulseaudio] alsa-util.c: Device surround21:1 doesn't support 44100 Hz, changed to 48000 Hz. ( 0.090| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-21 supported.

1 0

[alsa-devel] [asoc:topic/tas2552 1/8] sound/soc/codecs/tas2552.c:493:27-28: Unneeded semicolon
by kbuild test robot 05 Jun '15

05 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/tas2552 head: 16bd395259cf3e9966d40478891e0e610da109d4 commit: ea178d1456dcf88875d5edd148f2df8ea0de1794 [1/8] ASoC: tas2552: Make the enable-gpio really optional coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2552.c:493:27-28: Unneeded semicolon vim +493 sound/soc/codecs/tas2552.c 477 { 478 struct device *dev; 479 struct tas2552_data *data; 480 int ret; 481 int i; 482 483 dev = &client->dev; 484 data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); 485 if (data == NULL) 486 return -ENOMEM; 487 488 data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); 489 if (IS_ERR(data->enable_gpio)) { 490 if (PTR_ERR(data->enable_gpio) == -EPROBE_DEFER) 491 return -EPROBE_DEFER; 492 > 493 data->enable_gpio = NULL;; 494 } 495 496 data->tas2552_client = client; 497 data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); 498 if (IS_ERR(data->regmap)) { 499 ret = PTR_ERR(data->regmap); 500 dev_err(&client->dev, "Failed to allocate register map: %d\n", 501 ret); --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

2 1

[alsa-devel] [asoc:topic/zx 2/8] sound/soc/zte/zx296702-i2s.c:428:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 05 Jun '15

05 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/zx head: 69ccc50231eca9a57dc06a5514103c0f17ef402e commit: e5d4cd87800ce12c356e7eb571e565e839ab3a90 [2/8] ASoC: zx: Add ZTE zx296702 I2S DAI driver coccinelle warnings: (new ones prefixed by >>) >> sound/soc/zte/zx296702-i2s.c:428:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

4 6

[alsa-devel] Reflections on implementing Web MIDI on top of ALSA seq
by Adam Goode 05 Jun '15

05 Jun '15

Hi, Chrome 43 is now in beta, with Web MIDI support. (http://blog.chromium.org/2015/04/chrome-43-beta-web-midi-and-upgrading.html) The Linux (and Chrome OS) version of Chrome uses ALSA seq to implement Web MIDI. I wanted to reflect a bit on some of the issues I discovered while implementing this new API on top of seq. This is a little bit long, so the tl;dr is: * ALSA seq worked pretty well for implementing Web MIDI, but I need a way to get the card number for a given client, before my assumptions about the kernel break. Here is a quick overview of Web MIDI (http://www.w3.org/TR/webmidi/): - provides hotplug events for port connect/disconnect - incoming events are timestamped - outgoing events can be timestamped for future delivery by the system - timestamps are high precision, and use a monotonic clock - the port abstraction contains "name", "manufacturer", and "version" fields - open ports auto-reconnect if the port disappears and reappears, for both send and receive - input and output ports are distinct: there are no duplex ports - uses MIDI bytestreams (not an event structure), but requires full MIDI messages to be sent The initial prototype of Chrome's Web MIDI support was built on top of rawmidi. I rewrote it to use seq instead (to get multiple client and user client access). Here are my observations: - Getting the "manufacturer" is very hard from seq. I needed to join against udev data, which relies on the current implementation of ALSA in the kernel. These udev attributes are also used to match for transparent reconnect. - I could not use the kernel seq timestamp, either for send or receive, since I could not get it to just use a CLOCK_MONOTONIC offset. Instead, timestamping is done in userspace. - Hotplug was tricky, since I needed to listen to both seq events and udev events. I would prefer just to use udev, since I have to anyway. I definitely want to fix the kernel so that I can stop relying on a few implementation details. Here are the current assumptions I am making: - Card based clients start at 16. (I need to know which clients actually have cards) - Clients are in the same order as cards. - There is a 1:1 correspondence between clients and OPL3+rawmidi devices. One way to fix these assumptions would be to fully expose all seq events through sysfs. Right now, there is a seq device in sysfs but it has no information. Here are some ways that the kernel might be extended to simplify the code and/or fix bad assumptions: 1. Expose all seq clients+ports as kobjects, with a symlink to the card if it's a hardware device 2. Expose all subscriptions through sysfs 3. Add a simplified timestamp mechanism that just uses a configurable system clock, with at least CLOCK_MONOTONIC available (something like what sound/core/timer.c does with tread?) The first change would allow me to eliminate all the hardcoded assumptions about which cards correspond to which clients. I believe it would also allow me to drive the hotplug logic directly from udev, instead of both udev and seq. The second change is not strictly necessary, but would remove more need to react to seq events for some uses. The third change would allow me to use the kernel timestamp reliably, since right now I timestamp events with CLOCK_MONOTONIC in userspace. If I had to start today and create a new MIDI kernel interface for ALSA, it would look like this: not-quite-raw-midi: - hotplug events completely driven by sysfs - ability to interoperate with existing seq clients (for kernel and user clients) - non-exclusive access - reads and writes use a simple structured format, with timestamp and a midi message as bytes - no or minimal alsa-lib needed - one device file per port (no subdevices) - some mechanism for creating userspace clients, preferably with some number of key-value attributes (so I can report "manufacturer" and other things if the user client sets it) Thanks for reading these reflections. Comments are welcome. Adam

2 2

[alsa-devel] [PATCH] ASoC: zte: Fix missing dev in devm operation
by Mark Brown 04 Jun '15

04 Jun '15

Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/zte/zx296702-spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/zte/zx296702-spdif.c b/sound/soc/zte/zx296702-spdif.c index 4a93bca232ff..11a0e46a1156 100644 --- a/sound/soc/zte/zx296702-spdif.c +++ b/sound/soc/zte/zx296702-spdif.c @@ -309,7 +309,7 @@ static int zx_spdif_probe(struct platform_device *pdev) struct zx_spdif_info *zx_spdif; int ret; - zx_spdif = devm_kzalloc(sizeof(*zx_spdif), GFP_KERNEL); + zx_spdif = devm_kzalloc(&pdev->dev, sizeof(*zx_spdif), GFP_KERNEL); if (!zx_spdif) return -ENOMEM; -- 2.1.4

1 0

[alsa-devel] [PATCH v5 0/4] Support ZTE zx296702 SPDIF/I2S
by Jun Nie 04 Jun '15

04 Jun '15

Changes vs v4: - Fix devm_zalloc argument issue in SPDIF driver and static issue in i2s. Changes vs v3: - remove HDMI stuff and will add it in video side later - Polish SPDIF driver with devm_* Changes vs v2: - remove pcm dma driver and merge the related call into DAI driver - Add zx HDMI codec with direct HDMI function call - Other typo and minor fix Jun Nie (4): dt: Add documentation for the ZTE SPDIF controller dt: Add documentation for the ZTE I2S controller ASoC: zx: Add zx296702 SPDIF support ASoC: zx: Add ZTE zx296702 I2S DAI driver .../devicetree/bindings/sound/zte,zx-i2s.txt | 44 +++ .../devicetree/bindings/sound/zte,zx-spdif.txt | 28 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/zte/Kconfig | 17 + sound/soc/zte/Makefile | 2 + sound/soc/zte/zx296702-i2s.c | 437 +++++++++++++++++++++ sound/soc/zte/zx296702-spdif.c | 370 +++++++++++++++++ 8 files changed, 900 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-spdif.txt create mode 100644 sound/soc/zte/Kconfig create mode 100644 sound/soc/zte/Makefile create mode 100644 sound/soc/zte/zx296702-i2s.c create mode 100644 sound/soc/zte/zx296702-spdif.c -- 1.9.1

2 5

[alsa-devel] Different codecs for playback and capture?
by Ricard Wanderlof 04 Jun '15

04 Jun '15

Hi, I'm developing a machine driver for a system where we have two different codecs for input and output, connected to the same I2S interface. Normally a machine driver specifies which codec the I2S interface is connected to, but in this case there should really be two different ones, one for record and one for playback. I'm working on how to put this together, and just thought I'd try and get some input before going too far down the wrong track. The options I see are: 1. Specify two different codecs in the machine driver. Since one can specify several dai_links this would seem to be doable, however, I'm not sure ALSA can handle two codecs sharing a single CPU DAI in this way? 2. Use two separate machine drivers, one for each codec? Again, sharing a single CPU DAI would seem to be an issue here? 3. Write a special codec driver that is a merge between the two codecs, one for input and one for output. Seems like a bit of a hack, but should be possible. Any spontaneous thoughts? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 8

[alsa-devel] [PATCH v4 0/4] Support ZTE zx296702 SPDIF/I2S
by Jun Nie 04 Jun '15

04 Jun '15

Changes vs v3: - remove HDMI stuff and will add it in video side later - Polish SPDIF driver with devm_* Changes vs v2: - remove pcm dma driver and merge the related call into DAI driver - Add zx HDMI codec with direct HDMI function call - Other typo and minor fix Jun Nie (4): dt: Add documentation for the ZTE SPDIF controller dt: Add documentation for the ZTE I2S controller ASoC: zx: Add zx296702 SPDIF support ASoC: zx: Add ZTE zx296702 I2S DAI driver .../devicetree/bindings/sound/zte,zx-i2s.txt | 44 +++ .../devicetree/bindings/sound/zte,zx-spdif.txt | 28 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/zte/Kconfig | 17 + sound/soc/zte/Makefile | 2 + sound/soc/zte/zx296702-i2s.c | 437 +++++++++++++++++++++ sound/soc/zte/zx296702-spdif.c | 370 +++++++++++++++++ 8 files changed, 900 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-spdif.txt create mode 100644 sound/soc/zte/Kconfig create mode 100644 sound/soc/zte/Makefile create mode 100644 sound/soc/zte/zx296702-i2s.c create mode 100644 sound/soc/zte/zx296702-spdif.c -- 1.9.1

2 7

[alsa-devel] standalone dmic with omap-dmic
by Jack Mitchell 04 Jun '15

04 Jun '15

I have been trying to get a dmic connected directly into an omap44xx working standalone (no twl6040), so far, unsucessfully. Does anyone know if this is even possible? I have tried with a simple-audio-card setup using the dmic-codec and omap-dmic as the dai. I have also tried hacking out the twl6040 parts of the twl6040 driver to get a dmic only driver. With both of these methods I managed to get the drivers loaded but could get nothing recorded with an Input/Output error, and then on retry a kernel lockup with regards to enabling/disablig the dmic. Doe anyone have any suggestions; or maybe an idea if this is even possible without large changes? Does the dmic driver with twl6040 even work anymore; I expect it doesn't get a lot of testing... Cheers, Jack.

1 0

Re: [alsa-devel] [WM5102/ARIZONA] spi driver needs to support ACPI
by Pierre-Louis Bossart 04 Jun '15

04 Jun '15

On 6/3/15 2:50 AM, Christian Hartmann wrote: > hi, > > thanks Pierre-Luis, > > I attach the DSDT table in ASM format to get this device working as > soon as possible. this format is unreadable, can you extract the plain DSL instead, e.g. with iasl -d dsdt.dat -> extracts dsdt.dsl with only the code in 'human-readable' format. > > By the way, I looked yesterday into the baytrail code (sound/soc/intel/sst ) > and I added this WM510205 codec.... but I do not know if I have it > done correctly... currently I test my latest changes, seems to not > work as expected yet. > > > cheers > chris > > > 2015-06-02 20:13 GMT+02:00 Pierre-Louis Bossart > <pierre-louis.bossart(a)linux.intel.com>: >> On 06/02/2015 03:34 AM, Christian Hartmann wrote: >>> hi list again, >>> >>> I have patched the torvalds/master branch currently rc6 >>> and I have no NULL pointer exception anymore. >>> >>> I will send each patch separately >>> >>> Of course no (working) soundcard yet.. >>> >>> so let be try to describe what is actually open or resolved for me: >>> >>> * the spi master 37622 (pxa2xx) does not fail to add the SPI slave spi-WM510205 >>> pxa2xx-spi 80860F0E:00: registered child spi-WM510205:00 >>> >>> * ALSA device list: no soundcards found >>> * in dmesg I found two message indicating that something seems to be >>> missing yet: >>> >>> sst-acpi 80860F28:00: No matching ASoC machine driver found >>> intel-sst-acpi 80860F28:00: No matching machine driver found >>> >>> >>> snip of dmesg 4.1.0.42-rc6 >>> >>> >>> [ 6.075170] sst-acpi 80860F28:00: No matching ASoC machine driver found >>> [ 6.106787] pxa2xx-spi 80860F0E:00: no DMA channels available, using PIO >>> [ 6.106865] pxa2xx-spi 80860F0E:00: registered master spi32766 (dynamic) >>> [ 6.107058] spi spi-WM510205:00: 8333333 Hz actual, PIO >>> [ 6.107065] spi spi-WM510205:00: setup mode 0, 8 bits/w, 8000000 Hz max --> 0 >>> [ 6.107151] pxa2xx-spi 80860F0E:00: registered child spi-WM510205:00 >>> [ 6.114510] rfkill_gpio LNV4752:00: GPIO lookup for consumer reset >>> [ 6.114519] rfkill_gpio LNV4752:00: using ACPI for GPIO lookup >>> [ 6.114526] acpi LNV4752:00: GPIO: looking up reset-gpios >>> [ 6.114533] acpi LNV4752:00: GPIO: _DSD returned LNV4752:00 3 0 0 0 >>> [ 6.114607] no flags found for reset >>> [ 6.114619] rfkill_gpio LNV4752:00: GPIO lookup for consumer shutdown >>> [ 6.114623] rfkill_gpio LNV4752:00: using ACPI for GPIO lookup >>> [ 6.114628] acpi LNV4752:00: GPIO: looking up shutdown-gpios >>> [ 6.114633] acpi LNV4752:00: GPIO: _DSD returned LNV4752:00 3 1 0 0 >>> [ 6.114669] acpi LNV4752:00: GPIO: looking up shutdown-gpio >>> [ 6.114675] acpi LNV4752:00: GPIO: looking up 0 in _CRS >>> [ 6.114711] gpio-411 (reset): gpiod_request: status -16 >>> [ 6.114849] rfkill_gpio LNV4752:00: LNV4752:00 device registered. >>> [ 6.247784] intel_sst_acpi 80860F28:00: No matching machine driver found >>> >>> >>> I have grepped the string 80860F28 in the ACPI DSDT table (-> >>> dsdt.dsl) and found >>> >>> { >>> Device (LPEA) >>> { >>> Name (_ADR, Zero) // _ADR: Address >>> Name (_HID, "80860F28" /* Intel SST Audio DSP */) // >>> _HID: Hardware ID >>> Name (_CID, "80860F28" /* Intel SST Audio DSP */) // >>> _CID: Compatible ID >>> Name (_DDN, "Intel(R) Low Power Audio Controller - >>> 80860F28") // _DDN: DOS Device Name >>> Name (_SUB, "17AA7004") // _SUB: Subsystem ID >>> Name (_UID, One) // _UID: Unique ID >>> Name (_DEP, Package (0x01) // _DEP: Dependencies >>> { >>> ^SPI1.AUDI >>> }) >>> Name (_PR0, Package (0x01) // _PR0: Power Resources for D0 >>> { >>> PLPE >>> }) >>> Method (_STA, 0, NotSerialized) // _STA: Status >>> { >>> If (LAnd (LEqual (LPEE, 0x02), LEqual (LPED, Zero))) >>> { >>> Return (0x0F) >>> } >>> >>> Return (Zero) >>> } >>> >>> So how to bind the intel_sst_acpi with the WM5102 arizona coded and >>> its hub named arizone ??? >>> >>> Any suggestions or new directions? >> >> In your DSDT table, you need to leave the LPEA device alone. this is Intel only >> >> You will need to have a second device for your codec that is either a child of the I2C (or SPI?) controllers. This second device will have an HID that you need to include in the sst-acpi.c code. when the LPEA driver starts, it will look if this codec HID is present and if yes load the relevant machine driver and firmware. >> >> see for example the baytrail code, it uses the codec ID to identify which machine driver is needed. this is where you would need to add the wolfson HID. >> >> static struct sst_acpi_mach baytrail_machines[] = { >> { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, >> { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, >> {} >> }; >> >> >> Maybe you can share your DSDT table (factory settings) if you want more help. >> Also note that HIDs are specified based on rules defined by uefi.org, the "WM5102" does not follow any of the usual conventions. My guess is that the HID should be prefixed by the Wolfson PCI vendor ID. >> >> >>> .... >>> >>> >>> cheers >>> chris >>> >>> >>> 2015-05-19 15:39 GMT+02:00 Christian Hartmann <cornogle(a)googlemail.com>: >>>> Hi, >>>> >>>> at the moment I am totally confused. I thought adding an ACPI ID is >>>> trivial, but it is not. >>>> >>>> I have again a null pointer exception in arizona_spi_probe() >>>> >>>> could it be with the new num_chipselect value in spi-pxa2xx.c ?? >>>> I attach the trace of the last try >>>> >>>> Any hints or tips are welcome. >>>> >>>> cheers >>>> chris >>>> >>>> 2015-05-19 11:04 GMT+02:00 Christian Hartmann <cornogle(a)googlemail.com>: >>>>> Hi, >>>>> >>>>> so the ACPI id string, which I want to add to arizona-spi.c should be >>>>> >>>>> WM510205 ?? >>>>> >>>>> I have changed the ACPI string to >>>>> >>>>> {"WM510205", WM5102}, >>>>> >>>>> now and give it a try. >>>>> >>>>> cheers >>>>> chris >>>>> >>>>> 2015-05-18 18:29 GMT+02:00 Pierre-Louis Bossart >>>>> <pierre-louis.bossart(a)linux.intel.com>: >>>>>> On 5/15/15 4:16 AM, Christian Hartmann wrote: >>>>>>> >>>>>>> Hi list, >>>>>>> >>>>>>> >>>>>>> I need ACPI support for the arizona-core / arizona-spi driver for the >>>>>>> ACPI ID WM5102. >>>>>>> This audio codec is build into the Lenovo Yoga 851F. This device is a >>>>>>> 32Bit (UEFI) system >>>>>>> >>>>>>> >>>>>>> http://support.lenovo.com/de/de/products/tablets/yoga-series/yoga-tablet-2-… >>>>>>> >>>>>>> >>>>>>> I have tested with vanilla kernels from kernel.org 3.19.y ,4.0.y and >>>>>>> master (from torvalds) currently its 4.1.0-rc3. As distribution I am >>>>>>> working with Fedora / Fedlet was the initial install image. >>>>>>> >>>>>>> With all vanilla kernels I got at first an error message from spi32766: >>>>>>> >>>>>>> [ 0.296297] pxa2xx-spi 80860F0E:00: cs1 >= max 1 >>>>>>> [ 0.296311] spi_master spi32766: failed to add SPI device >>>>>>> WM510205:00 from ACPI >>>>>>> >>>>>>> so I have at first patched as suggested by broonie the arizona-core >>>>>>> >>>>>>> commit 48203a0fede057d6a8c5f6872c53326395350f45 >>>>>>> Author: somebody anybody <somebody(a)foobar.com> >>>>>>> Date: Mon May 11 13:35:43 2015 +0200 >>>>>>> >>>>>>> spi-pxa2xx : raise chipselect >>>>>>> >>>>>>> Signed-off-by: somebody anybody <somebody(a)foobar.com> >>>>>>> >>>>>>> diff --git a/drivers/spi/spi-pxa2xx.c b/drivers/spi/spi-pxa2xx.c >>>>>>> index e3223ac..9c919ea 100644 >>>>>>> --- a/drivers/spi/spi-pxa2xx.c >>>>>>> +++ b/drivers/spi/spi-pxa2xx.c >>>>>>> @@ -1279,7 +1279,7 @@ pxa2xx_spi_acpi_get_pdata(struct platform_device >>>>>>> *pdev) >>>>>>> if (adev->pnp.unique_id && !kstrtoint(adev->pnp.unique_id, 0, >>>>>>> &devid)) >>>>>>> ssp->port_id = devid; >>>>>>> >>>>>>> - pdata->num_chipselect = 1; >>>>>>> + pdata->num_chipselect = 20; >>>>>>> pdata->enable_dma = true; >>>>>>> >>>>>>> return pdata; >>>>>>> >>>>>>> This leds the spi master do the further ACPI configuration and inital >>>>>>> setup for the arizona-spi device, but adding the ACPI ID to it >>>>>>> does not work yet as expected: >>>>>>> >>>>>>> The current last patchset on top of master is >>>>>>> >>>>>>> >>>>>>> Author: somebody anybody <somebody(a)foobar.com> >>>>>>> Date: Wed May 13 11:36:25 2015 +0200 >>>>>>> >>>>>>> arizona : added ACPI id of wm5102 to SPI driver, too >>>>>>> >>>>>>> Signed-off-by: somebody anybody <somebody(a)foobar.com> >>>>>>> >>>>>>> diff --git a/drivers/mfd/arizona-spi.c b/drivers/mfd/arizona-spi.c >>>>>>> index 1e845f6..c19a4c1 100644 >>>>>>> --- a/drivers/mfd/arizona-spi.c >>>>>>> +++ b/drivers/mfd/arizona-spi.c >>>>>>> @@ -17,6 +17,7 @@ >>>>>>> #include <linux/regulator/consumer.h> >>>>>>> #include <linux/slab.h> >>>>>>> #include <linux/spi/spi.h> >>>>>>> +#include <linux/acpi.h> >>>>>>> #include <linux/of.h> >>>>>>> >>>>>>> #include <linux/mfd/arizona/core.h> >>>>>>> @@ -82,6 +83,15 @@ static int arizona_spi_remove(struct spi_device *spi) >>>>>>> return 0; >>>>>>> } >>>>>>> >>>>>>> +#ifdef CONFIG_ACPI >>>>>>> +static struct acpi_device_id wm5102_acpi_match[] = { >>>>>>> + { "WM5102", 0}, >>>>>>> + { "WM510205", 0}, >>>>>>> + { }, >>>>>>> +}; >>>>>>> +MODULE_DEVICE_TABLE(acpi, wm5102_acpi_match); >>>>>>> +#endif >>>>>>> + >>>>>>> static const struct spi_device_id arizona_spi_ids[] = { >>>>>>> { "wm5102", WM5102 }, >>>>>>> { "wm5110", WM5110 }, >>>>>>> @@ -96,6 +106,9 @@ static struct spi_driver arizona_spi_driver = { >>>>>>> .owner = THIS_MODULE, >>>>>>> .pm = &arizona_pm_ops, >>>>>>> .of_match_table = of_match_ptr(arizona_of_match), >>>>>>> +#ifdef CONFIG_ACPI >>>>>>> + .acpi_match_table = ACPI_PTR(wm5102_acpi_match), >>>>>>> +#endif >>>>>>> }, >>>>>>> .probe = arizona_spi_probe, >>>>>>> .remove = arizona_spi_remove, >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> commit 268e21d29c942205e40f749c6d71c77839dc67da >>>>>>> Author: somebody anybody <somebody(a)foobar.com> >>>>>>> Date: Wed May 13 11:36:04 2015 +0200 >>>>>>> >>>>>>> arizona : added ACPI id of wm5102 to i2c driver >>>>>>> >>>>>>> Signed-off-by: somebody anybody <somebody(a)foobar.com> >>>>>>> >>>>>>> diff --git a/drivers/mfd/arizona-i2c.c b/drivers/mfd/arizona-i2c.c >>>>>>> index ff782a5..45ece1b 100644 >>>>>>> --- a/drivers/mfd/arizona-i2c.c >>>>>>> +++ b/drivers/mfd/arizona-i2c.c >>>>>>> @@ -18,6 +18,7 @@ >>>>>>> #include <linux/regulator/consumer.h> >>>>>>> #include <linux/slab.h> >>>>>>> #include <linux/of.h> >>>>>>> +#include <linux/acpi.h> >>>>>>> >>>>>>> #include <linux/mfd/arizona/core.h> >>>>>>> >>>>>>> @@ -85,6 +86,15 @@ static int arizona_i2c_remove(struct i2c_client *i2c) >>>>>>> return 0; >>>>>>> } >>>>>>> >>>>>>> +#ifdef CONFIG_ACPI >>>>>>> +static struct acpi_device_id wm5102_acpi_match[] = { >>>>>>> + { "WM5102", 0}, >>>>>>> + { "WM510205", 0}, >>>>>>> + { }, >>>>>> >>>>>> >>>>>> Those IDs don't really look like ACPI ones? Isn't there a rule that the _HID >>>>>> is represented by a 4-letter vendor ID followed by part ID? see >>>>>> http://www.uefi.org/PNP_ACPI_Registry >>>>>> >>>>>> >>>>>>> +}; >>>>>>> +MODULE_DEVICE_TABLE(acpi, wm5102_acpi_match); >>>>>>> +#endif >>>>>>> + >>>>>>> static const struct i2c_device_id arizona_i2c_id[] = { >>>>>>> { "wm5102", WM5102 }, >>>>>>> { "wm5110", WM5110 }, >>>>>>> @@ -100,6 +110,9 @@ static struct i2c_driver arizona_i2c_driver = { >>>>>>> .owner = THIS_MODULE, >>>>>>> .pm = &arizona_pm_ops, >>>>>>> .of_match_table = of_match_ptr(arizona_of_match), >>>>>>> +#ifdef CONFIG_ACPI >>>>>>> + .acpi_match_table = ACPI_PTR(wm5102_acpi_match), >>>>>>> +#endif >>>>>>> }, >>>>>>> .probe = arizona_i2c_probe, >>>>>>> .remove = arizona_i2c_remove, >>>>>>> >>>>>>> >>>>>>> >>>>>>> with these two patches the local master branch compiles fine, but at >>>>>>> initialzing the arizona-spi driver gots a NULL pointer exception. >>>>>>> >>>>>>> I attach the whole dmesg of 4.1.0-rc3 (build 4.1.0.24-rc3). >>>>>>> >>>>>>> I need some assistance or hints to the right direction, I am looking >>>>>>> myself and I also try what I can, but I think it would be easy peasy >>>>>>> with some experts like here on this list :) >>>>>>> >>>>>>> >>>>>>> PS: the message : spi_master spi32766: raised num_chipselect to 20 for >>>>>>> WM510205:00 >>>>>>> >>>>>>> is a dev_err() line added to the function static acpi_status >>>>>>> acpi_spi_add_device(), there I had raised the num_chipselect first >>>>>>> (that was wrong, the raise is now done as in the first patch above. >>>>>>> this is only a debug message and will be removed soon) >>>>>>> >>>>>>> >>>>>>> PPS: for me it seems that there is a missing 'link' from WM5102 to the >>>>>>> sst-acpi 80860F28:00 device, which complaints NOW / the first time >>>>>>> with the three patches applied on top on master.... will have a look >>>>>>> into it. >>>>>>> >>>>>>> I hope I got this device working with some more help from you. >>>>>>> >>>>>>> cheers >>>>>>> chris >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Alsa-devel mailing list >>>>>>> Alsa-devel(a)alsa-project.org >>>>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>>>> >>>>> >>> _______________________________________________ >>> Alsa-devel mailing list >>> Alsa-devel(a)alsa-project.org >>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>> >> > > >

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[alsa-devel] Still have problems/issues using SGTL5000 with i.MX28 CPU
by gianluca 04 Jun '15

04 Jun '15

Hello everybody! I put all those people on Cc: because of I think every guy can be able to handle the issue for its side easly. I started a discussion in FreeScale Community website: https://community.freescale.com/thread/356362 Basically I designed 4 boards using the IMX28EVK board as reference. Usually the audio section was a buzzer (pwm), or simply using a SGTL5000 as playback only. Now we need to record some sounds as soon as the SGTL5000 has the SAIF1 line used for caputure (line in / mic in). In alsamixer I can see all those stuff, and actually it works (for example if I bypass the ADC part to listen any MIC or LINE-IN into headphones directly) so I can safely exclude some alsa misprogramming code or whatever. I tested my bootlets, barebox bootloader, linux kernel 3.12.1 and Debian Wheezy 7.8 in both boards: * my board (namely EK340) * IMX28VK board. Both bootcodes and barebox bootloader are almost the same in both boards (some uart definitions differ). The linux kernel _IS_ the same. The device-tree blob are quite similar: Here is the IMX28EVK dts: > /* > * Copyright 2012 Freescale Semiconductor, Inc. > * > * The code contained herein is licensed under the GNU General Public > * License. You may obtain a copy of the GNU General Public License > * Version 2 or later at the following locations: > * > * http://www.opensource.org/licenses/gpl-license.html > * http://www.gnu.org/copyleft/gpl.html > */ > > /dts-v1/; > #include "imx28.dtsi" > > / { > /* > * Try to keep the name of the board unique (no slash, underscore, > * no fancy characters, no spaces, no dots...) in the second > * field of the model name. It will be used as board name. > */ > model = "Freescale IMX28EVK Evaluation Kit"; > compatible = "fsl,imx28-evk", "fsl,imx28"; > > memory { > reg = <0x40000000 0x08000000>; > }; > > apb@80000000 { > apbh@80000000 { > ssp0: ssp@80010000 { > compatible = "fsl,imx28-mmc"; > pinctrl-names = "default"; > pinctrl-0 = <&mmc0_8bit_pins > &mmc0_sck_cfg>; > bus-width = <8>; > wp-gpios = <&gpio2 12 0>; > vmmc-supply = <&reg_vddio_sd0>; > non-removable; > status = "okay"; > }; > > ssp1: ssp@80012000 { > compatible = "fsl,imx28-mmc"; > pinctrl-names = "default"; > pinctrl-0 = <&mmc1_4bit_pins > &mmc1_cd_cfg &mmc1_sck_cfg>; > bus-width = <4>; > wp-gpios = <&gpio0 28 0>; > vmmc-supply = <&reg_vddio_sd0>; > status = "okay"; > }; > > pinctrl@80018000 { > pinctrl-names = "default"; > pinctrl-0 = <&hog_pins_a>; > > hog_pins_a: hog@0 { > reg = <0>; > fsl,pinmux-ids = < > 0x20d3 /* MX28_PAD_SSP1_CMD__GPIO_2_13 */ > 0x20f3 /* MX28_PAD_SSP1_DATA3__GPIO_2_15 */ > 0x40d3 /* MX28_PAD_ENET0_RX_CLK__GPIO_4_13 */ > 0x20c3 /* MX28_PAD_SSP1_SCK__GPIO_2_12 */ > 0x31c3 /* MX28_PAD_PWM3__GPIO_3_28 */ > 0x31e3 /* MX28_PAD_LCD_RESET__GPIO_3_30 */ > 0x3083 /* MX28_PAD_AUART2_RX__GPIO_3_8 */ > 0x3093 /* MX28_PAD_AUART2_TX__GPIO_3_9 */ > >; > fsl,drive-strength = <0>; > fsl,voltage = <1>; > fsl,pull-up = <0>; > }; > > rs485_tx_en: rx485-tx-en { > fsl,pinmux-ids = < > 0x3053 /* MX28_PAD_AUART1_TX__GPIO_3_5 */ > >; > fsl,drive-strength = <0>; > fsl,pull-up = <0>; > }; > > mmc0_8bit_pins: mmc0-8bit@0 { > reg = <0>; > fsl,pinmux-ids = < > 0x2000 /* MX28_PAD_SSP0_DATA0__SSP0_D0 */ > 0x2010 /* MX28_PAD_SSP0_DATA1__SSP0_D1 */ > 0x2020 /* MX28_PAD_SSP0_DATA2__SSP0_D2 */ > 0x2030 /* MX28_PAD_SSP0_DATA3__SSP0_D3 */ > 0x2040 /* MX28_PAD_SSP0_DATA4__SSP0_D4 */ > 0x2050 /* MX28_PAD_SSP0_DATA5__SSP0_D5 */ > 0x2060 /* MX28_PAD_SSP0_DATA6__SSP0_D6 */ > 0x2070 /* MX28_PAD_SSP0_DATA7__SSP0_D7 */ > 0x2080 /* MX28_PAD_SSP0_CMD__SSP0_CMD */ > 0x20a0 /* MX28_PAD_SSP0_SCK__SSP0_SCK */ > >; > fsl,drive-strength = <1>; > fsl,voltage = <1>; > fsl,pull-up = <1>; > }; > > mmc1_4bit_pins: mmc1-4bit@0 { > reg = <0>; > fsl,pinmux-ids = < > 0x0001 /* MX28_PAD_GPMI_D00__SSP1_D0 */ > 0x0011 /* MX28_PAD_GPMI_D01__SSP1_D1 */ > 0x0021 /* MX28_PAD_GPMI_D02__SSP1_D2 */ > 0x0031 /* MX28_PAD_GPMI_D03__SSP1_D3 */ > 0x0141 /* MX28_PAD_GPMI_RDY0__SSP1_CARD_DETECT */ > 0x0151 /* MX28_PAD_GPMI_RDY1__SSP1_CMD */ > 0x0191 /* MX28_PAD_GPMI_WRN__SSP1_SCK */ > >; > fsl,drive-strength = <1>; > fsl,voltage = <1>; > fsl,pull-up = <1>; > }; > > mmc1_sck_cfg: mmc1-sck-cfg { > fsl,pinmux-ids = < > 0x0191 /* MX28_PAD_GPMI_WRN__SSP1_SCK */ > >; > fsl,drive-strength = <1>; > fsl,pull-up = <0>; > }; > > mmc1_cd_cfg: mmc1-cd-cfg { > fsl,pinmux-ids = < > 0x0141 /* MX28_PAD_GPMI_RDY0__SSP1_CARD_DETECT */ > >; > fsl,pull-up = <0>; > }; > > lcdif_pins_evk: lcdif-evk@0 { > reg = <0>; > fsl,pinmux-ids = < > 0x1181 /* MX28_PAD_LCD_RD_E__LCD_VSYNC */ > 0x1191 /* MX28_PAD_LCD_WR_RWN__LCD_HSYNC */ > 0x11a1 /* MX28_PAD_LCD_RS__LCD_DOTCLK */ > 0x11b1 /* MX28_PAD_LCD_CS__LCD_ENABLE */ > >; > fsl,drive-strength = <0>; > fsl,voltage = <1>; > fsl,pull-up = <0>; > }; > > usb0_id_pins: usb0id@0 { > reg = <0>; > fsl,pinmux-ids = < > 0x3071 /* MX28_PAD_AUART1_RTS__USB0_ID */ > >; > fsl,drive-strength = <2>; > fsl,voltage = <1>; > fsl,pull-up = <1>; > }; > > }; > > lcdif@80030000 { > pinctrl-names = "default"; > pinctrl-0 = <&lcdif_24bit_pins_a > &lcdif_pins_evk>; > lcd-supply = <&reg_lcd_3v3>; > display = <&display>; > status = "okay"; > > display: display { > bits-per-pixel = <32>; > bus-width = <24>; > > display-timings { > native-mode = <&timing0>; > timing0: timing0 { > clock-frequency = <33500000>; > hactive = <800>; > vactive = <480>; > hback-porch = <89>; > hfront-porch = <164>; > vback-porch = <23>; > vfront-porch = <10>; > hsync-len = <10>; > vsync-len = <10>; > hsync-active = <0>; > vsync-active = <0>; > de-active = <1>; > pixelclk-active = <0>; > }; > }; > }; > }; > > can0: can@80032000 { > pinctrl-names = "default"; > pinctrl-0 = <&can0_pins_a>; > xceiver-supply = <&reg_can_3v3>; > status = "okay"; > }; > > can1: can@80034000 { > pinctrl-names = "default"; > pinctrl-0 = <&can1_pins_a>; > xceiver-supply = <&reg_can_3v3>; > status = "okay"; > }; > }; > > apbx@80040000 { > saif0: saif@80042000 { > pinctrl-names = "default"; > pinctrl-0 = <&saif0_pins_a>; > status = "okay"; > }; > > saif1: saif@80046000 { > pinctrl-names = "default"; > pinctrl-0 = <&saif1_pins_a>; > fsl,saif-master = <&saif0>; > status = "okay"; > }; > > lradc@80050000 { > fsl,lradc-touchscreen-wires = <4>; > status = "okay"; > }; > > i2c0: i2c@80058000 { > pinctrl-names = "default"; > pinctrl-0 = <&i2c0_pins_a>; > clock-frequency = <400000>; > status = "okay"; > > sgtl5000: codec@0a { > compatible = "fsl,sgtl5000"; > reg = <0x0a>; > VDDA-supply = <&reg_3p3v>; > VDDIO-supply = <&reg_3p3v>; > clocks = <&saif0>; > }; > > at24@51 { > compatible = "at24,24c32"; > pagesize = <32>; > reg = <0x51>; > }; > }; > > pwm: pwm@80064000 { > pinctrl-names = "default"; > pinctrl-0 = <&pwm2_pins_a>; > status = "okay"; > }; > > duart: serial@80074000 { > pinctrl-names = "default"; > pinctrl-0 = <&duart_pins_a>; > status = "okay"; > }; > > auart0: serial@8006a000 { > pinctrl-names = "default"; > pinctrl-0 = <&auart0_pins_a>; > fsl,uart-has-rtscts; > status = "okay"; > }; > > auart3: serial@80070000 { > compatible = "fsl,imx28-ekuart"; > pinctrl-names = "default"; > pinctrl-0 = <&auart3_2pins_b > &rs485_tx_en>; > rs485-rts-active-high; > rts-gpio = <&gpio3 5 0>; > linux,rs485-enabled-at-boot-time; > rs485-rts-delay = <0 0>; > status = "okay"; > }; > > usbphy0: usbphy@8007c000 { > status = "okay"; > }; > > usbphy1: usbphy@8007e000 { > status = "okay"; > }; > }; > }; > > ahb@80080000 { > usb0: usb@80080000 { > pinctrl-names = "default"; > pinctrl-0 = <&usb0_id_pins > &usbphy0_pins_b>; > vbus-supply = <&reg_usb0_vbus>; > dr_mode = "otg"; > status = "okay"; > }; > > usb1: usb@80090000 { > pinctrl-names = "default"; > pinctrl-0 = <&usbphy1_pins_a>; > vbus-supply = <&reg_usb1_vbus>; > dr_mode = "host"; > status = "okay"; > }; > > mac0: ethernet@800f0000 { > phy-mode = "rmii"; > pinctrl-names = "default"; > pinctrl-0 = <&mac0_pins_a>; > phy-supply = <&reg_fec_3v3>; > phy-reset-gpios = <&gpio4 13 0>; > phy-reset-duration = <100>; > status = "okay"; > }; > > mac1: ethernet@800f4000 { > phy-mode = "rmii"; > pinctrl-names = "default"; > pinctrl-0 = <&mac1_pins_a>; > status = "okay"; > }; > }; > > regulators { > compatible = "simple-bus"; > > reg_3p3v: 3p3v { > compatible = "regulator-fixed"; > regulator-name = "3P3V"; > regulator-min-microvolt = <3300000>; > regulator-max-microvolt = <3300000>; > regulator-always-on; > }; > > reg_vddio_sd0: vddio-sd0 { > compatible = "regulator-fixed"; > regulator-name = "vddio-sd0"; > regulator-min-microvolt = <3300000>; > regulator-max-microvolt = <3300000>; > gpio = <&gpio3 28 0>; > }; > > reg_fec_3v3: fec-3v3 { > compatible = "regulator-fixed"; > regulator-name = "fec-3v3"; > regulator-min-microvolt = <3300000>; > regulator-max-microvolt = <3300000>; > gpio = <&gpio2 15 0>; > }; > > reg_usb0_vbus: usb0_vbus { > compatible = "regulator-fixed"; > regulator-name = "usb0_vbus"; > regulator-min-microvolt = <5000000>; > regulator-max-microvolt = <5000000>; > gpio = <&gpio3 9 0>; > enable-active-high; > }; > > reg_usb1_vbus: usb1_vbus { > compatible = "regulator-fixed"; > regulator-name = "usb1_vbus"; > regulator-min-microvolt = <5000000>; > regulator-max-microvolt = <5000000>; > gpio = <&gpio3 8 0>; > enable-active-high; > }; > > reg_lcd_3v3: lcd-3v3 { > compatible = "regulator-fixed"; > regulator-name = "lcd-3v3"; > regulator-min-microvolt = <3300000>; > regulator-max-microvolt = <3300000>; > gpio = <&gpio3 30 0>; > enable-active-high; > }; > > reg_can_3v3: can-3v3 { > compatible = "regulator-fixed"; > regulator-name = "can-3v3"; > regulator-min-microvolt = <3300000>; > regulator-max-microvolt = <3300000>; > gpio = <&gpio2 13 0>; > enable-active-high; > }; > > }; > > sound { > compatible = "fsl,imx28-evk-sgtl5000", > "fsl,mxs-audio-sgtl5000"; > model = "imx28-evk-sgtl5000"; > saif-controllers = <&saif0 &saif1>; > audio-codec = <&sgtl5000>; > }; > > backlight { > compatible = "pwm-backlight"; > pwms = <&pwm 2 5000000>; > brightness-levels = <0 4 8 16 32 64 128 255>; > default-brightness-level = <6>; > }; > }; and here the EK340 dts: > /* > * Copyright 2014 Eurek Elettronica S.r.l. > * > * The code contained herein is licensed under the GNU General Public > * License. You may obtain a copy of the GNU General Public License > * Version 2 or later at the following locations: > * > * http://www.opensource.org/licenses/gpl-license.html > * http://www.gnu.org/copyleft/gpl.html > */ > > /dts-v1/; > #include "imx28.dtsi" > > / { > model = "Eurek EK340 i.MX286"; > compatible = "eurek,ek340", "fsl,imx28"; > > memory { > reg = <0x40000000 0x08000000>; > }; > > apb@80000000 { > apbh@80000000 { > ssp0: ssp@80010000 { > compatible = "fsl,imx28-mmc"; > pinctrl-names = "default"; > pinctrl-0 = <&mmc0_8bit_pins > &mmc0_sck_cfg>; > bus-width = <8>; > vmmc-supply = <&reg_3p3v>; > non-removable; > status = "okay"; > }; > > ssp1: ssp@80012000 { > compatible = "fsl,imx28-mmc"; > pinctrl-names = "default"; > pinctrl-0 = <&mmc1_4bit_pins > &mmc1_cd_cfg &mmc1_sck_cfg>; > bus-width = <4>; > vmmc-supply = <&reg_vddio_sd1>; > status = "okay"; > }; > > ssp3: ssp@80016000 { > compatible = "fsl,imx28-sdio"; > pinctrl-names = "default"; > pinctrl-0 = <&mmc3_4bit_pins > &mmc3_sck_cfg>; > bus-width = <4>; > non-removable; > max-frequency = <25000000>; > status = "okay"; > }; > > pinctrl@80018000 { > pinctrl-names = "default"; > pinctrl-0 = <&hog_pins_a>; > > hog_pins_a: hog@0 { > reg = <0>; > fsl,pinmux-ids = < > 0x0073 /* MX28_PAD_GPMI_D07__GPIO_0_7 LED */ > 0x0063 /* MX28_PAD_GPMI_D06__GPIO_0_6 WIFI PWREN */ > 0x31e3 /* MX28_PAD_LCD_RESET__GPIO_3_30 FACT_RST */ > 0x2093 /* MX28_PAD_SSP0_DETECT__GPIO_2_9 MMC_nRST */ > 0x0043 /* MX28_PAD_GPMI_D04__GPIO_0_4 USB0_PWREN */ > 0x0123 /* MX28_PAD_GPMI_CE2N__GPIO_0_18 USB1_PWREN */ > 0x3013 /* MX28_PAD_AUART0_TX__GPIO_3_1 ENET_nRST */ > 0x3003 /* MX28_PAD_AUART0_RX__GPIO_3_0 ENET_INT */ > 0x0053 /* MX28_PAD_GPMI_D05__GPIO_0_5 LCD_PCI */ > 0x31b3 /* MX28_PAD_SPDIF__GPIO_3_27 BL_EN (I2C_INT) */ > 0x2123 /* MX28_PAD_SSP2_MISO__GPIO_2_18 EXT_INT */ > >; > fsl,drive-strength = <0>; > fsl,voltage = <1>; > fsl,pull-up = <0>; > }; > > rs485_tx_en: rx485-tx-en { > fsl,pinmux-ids = < > 0x2133 /* MX28_PAD_SSP2_SS0__GPIO_2_19 UART2_DIR */ > >; > fsl,drive-strength = <0>; > fsl,pull-up = <1>; > }; > > powerfail_pin: powerfail-pin { > fsl,pinmux-ids = < > 0x0133 /* MX28_PAD_GPMI_CE3N__GPIO_0_19 PWR_FAIL */ > >; > fsl,drive-strength = <0>; > fsl,pull-up = <0>; > }; > > mmc0_8bit_pins: mmc0-8bit@0 { > reg = <0>; > fsl,pinmux-ids = < > 0x2000 /* MX28_PAD_SSP0_DATA0__SSP0_D0 */ > 0x2010 /* MX28_PAD_SSP0_DATA1__SSP0_D1 */ > 0x2020 /* MX28_PAD_SSP0_DATA2__SSP0_D2 */ > 0x2030 /* MX28_PAD_SSP0_DATA3__SSP0_D3 */ > 0x2040 /* MX28_PAD_SSP0_DATA4__SSP0_D4 */ > 0x2050 /* MX28_PAD_SSP0_DATA5__SSP0_D5 */ > 0x2060 /* MX28_PAD_SSP0_DATA6__SSP0_D6 */ > 0x2070 /* MX28_PAD_SSP0_DATA7__SSP0_D7 */ > 0x2080 /* MX28_PAD_SSP0_CMD__SSP0_CMD */ > 0x20a0 /* MX28_PAD_SSP0_SCK__SSP0_SCK */ > >; > fsl,drive-strength = <1>; > fsl,voltage = <1>; > fsl,pull-up = <1>; > }; > > mmc1_4bit_pins: mmc1-4bit@0 { > reg = <0>; > fsl,pinmux-ids = < > 0x0001 /* MX28_PAD_GPMI_D00__SSP1_D0 */ > 0x0011 /* MX28_PAD_GPMI_D01__SSP1_D1 */ > 0x0021 /* MX28_PAD_GPMI_D02__SSP1_D2 */ > 0x0031 /* MX28_PAD_GPMI_D03__SSP1_D3 */ > 0x0141 /* MX28_PAD_GPMI_RDY0__SSP1_CARD_DETECT */ > 0x0151 /* MX28_PAD_GPMI_RDY1__SSP1_CMD */ > 0x0191 /* MX28_PAD_GPMI_WRN__SSP1_SCK */ > >; > fsl,drive-strength = <1>; > fsl,voltage = <1>; > fsl,pull-up = <1>; > }; > > mmc1_sck_cfg: mmc1-sck-cfg { > fsl,pinmux-ids = < > 0x0191 /* MX28_PAD_GPMI_WRN__SSP1_SCK */ > >; > fsl,drive-strength = <2>; > fsl,pull-up = <0>; > }; > > mmc1_cd_cfg: mmc1-cd-cfg { > fsl,pinmux-ids = < > 0x0141 /* MX28_PAD_GPMI_RDY0__SSP1_CARD_DETECT */ > >; > fsl,pull-up = <0>; > }; > > mmc3_4bit_pins: mmc3-4bit@0 { > reg = <0>; > fsl,pinmux-ids = < > 0x0101 /* MX28_PAD_GPMI_CE0N__SSP3_D0 */ > 0x01a1 /* MX28_PAD_GPMI_ALE__SSP3_D1 */ > 0x01b1 /* MX28_PAD_GPMI_CLE__SSP3_D2 */ > 0x0111 /* MX28_PAD_GPMI_CE1N__SSP3_D3 */ > 0x01c1 /* MX28_PAD_GPMI_RESETN__SSP3_CMD */ > 0x0181 /* MX28_PAD_GPMI_RDN__SSP3_SCK */ > >; > fsl,drive-strength = <1>; > fsl,voltage = <1>; > fsl,pull-up = <1>; > }; > > mmc3_sck_cfg: mmc3-sck-cfg { > fsl,pinmux-ids = < > 0x0181 /* MX28_PAD_GPMI_RDN__SSP3_SCK */ > >; > fsl,drive-strength = <2>; > fsl,pull-up = <0>; > }; > > usb0_id_pins_b: usb0_id@1 { > reg = <0>; > fsl,pinmux-ids = < > 0x3121 /* MX28_PAD_PWM2__USB0_ID */ > >; > fsl,drive-strength = <2>; > fsl,voltage = <1>; > fsl,pull-up = <1>; > }; > > lcdif_pins_ek340: lcdif-ek340@0 { > reg = <0>; > fsl,pinmux-ids = < > 0x1181 /* MX28_PAD_LCD_RD_E__LCD_VSYNC */ > 0x1191 /* MX28_PAD_LCD_WR_RWN__LCD_HSYNC */ > 0x11a1 /* MX28_PAD_LCD_RS__LCD_DOTCLK */ > 0x11b1 /* MX28_PAD_LCD_CS__LCD_ENABLE */ > >; > fsl,drive-strength = <0>; > fsl,voltage = <1>; > fsl,pull-up = <0>; > }; > }; > > lcdif@80030000 { > pinctrl-names = "default"; > pinctrl-0 = <&lcdif_24bit_pins_a > &lcdif_pins_ek340>; > lcd-supply = <&reg_lcd_3v3>; > > display = <&display16>; > status = "okay"; > > display16: display16 { > bits-per-pixel = <16>; > bus-width = <16>; > > display-timings { > native-mode = <&lms700kf23>; > > /* DISPLAY 800x480 SAMSUNG LMS700KF23 */ > lms700kf23: lms700kf23 { > clock-frequency = <24500000>; > hactive = <800>; > vactive = <480>; > hback-porch = <16>; > hfront-porch = <8>; > vback-porch = <8>; > vfront-porch = <5>; > hsync-len = <1>; > vsync-len = <1>; > hsync-active = <0>; > vsync-active = <0>; > de-active = <1>; > pixelclk-active = <0>; > }; > > /* DISPLAY 480x272 AMPIRE AM-480272Q */ > am480272q: am480272q { > clock-frequency = <9200000>; > hactive = <480>; > vactive = <272>; > hback-porch = <43>; > hfront-porch = <8>; > vback-porch = <12>; > vfront-porch = <4>; > hsync-len = <10>; > vsync-len = <41>; > hsync-active = <0>; > vsync-active = <0>; > de-active = <1>; > pixelclk-active = <0>; > }; > /* DISPLAY 1024x600 AMPIRE AM-1024600LTM LVDS */ > am1024600l: am1024600l { > clock-frequency = <51200000>; > hactive = <1024>; > vactive = <600>; > hback-porch = <0>; > hfront-porch = <320>; > vback-porch = <0>; > vfront-porch = <35>; > hsync-len = <1>; > vsync-len = <1>; > hsync-active = <0>; > vsync-active = <0>; > de-active = <1>; > pixelclk-active = <0>; > }; > }; > }; > > display24: display24 { > bits-per-pixel = <32>; > bus-width = <24>; > /* Timings for 24bit display go here... */ > }; > > }; > > can0: can@80032000 { > pinctrl-names = "default"; > pinctrl-0 = <&can0_pins_a>; > xceiver-supply = <&reg_3p3v>; > status = "okay"; > }; > }; > > > apbx@80040000 { > saif0: saif@80042000 { > pinctrl-names = "default"; > pinctrl-0 = <&saif0_pins_a>; > status = "okay"; > }; > > saif1: saif@80046000 { > pinctrl-names = "default"; > pinctrl-0 = <&saif1_pins_a>; > fsl,saif-master = <&saif0>; > status = "okay"; > }; > > lradc@80050000 { > fsl,lradc-touchscreen-wires = <4>; > status = "okay"; > }; > > i2c0: i2c@80058000 { > pinctrl-names = "default"; > pinctrl-0 = <&i2c0_pins_a>; > clock-frequency = <400000>; > status = "okay"; > > sgtl5000: codec@0a { > compatible = "fsl,sgtl5000"; > reg = <0x0a>; > VDDA-supply = <&reg_3p3v>; > VDDIO-supply = <&reg_3p3v>; > clocks = <&saif0>; > }; > > at24@50 { > compatible = "at24,24c64"; > pagesize = <64>; > reg = <0x50>; > }; > > pcf8563: rtc@51 { > compatible = "phg,pcf8563"; > reg = <0x51>; > }; > > /* > * ------------------------------------------------ > * There is no need to bind those devices to Linux > * Kernel space. They are usable with i2c tools and > * custom i2c userspace drivers anyway. > * !! Dangerous !! > * ------------------------------------------------ > pca9536: pca9536@41 { > compatible = "nxp,pca9536"; > gpio-controller; > reg = <0x41>; > }; > > at24@54 { > compatible = "at24,24c64"; > pagesize = <64>; > reg = <0x54>; > }; > > lm48100@7c { > compatible = "alsa-volume-ctrl"; > volume-controller; > reg = <0x7c>; > }; > * ------------------------------------------------ > */ > }; > > i2c1: i2c@8005a000 { > pinctrl-names = "default"; > pinctrl-0 = <&i2c1_pins_a>; > clock-frequency = <100000>; > status = "okay"; > }; > > pwm: pwm@80064000 { > compatible = "imx28-ek-pwm"; > reg = <0x80064000 0x2000>; > clocks = <&clks 44>; > #pwm-cells = <2>; > fsl,pwm-number = <8>; > pinctrl-names = "default"; > pinctrl-0 = <&pwm4_pins_a > &powerfail_pin>; > powerfail-active-low; > powerfail-gpio = <&gpio0 19 1>; > status = "okay"; > }; > > duart: serial@80074000 { > pinctrl-names = "default"; > pinctrl-0 = <&duart_pins_b>; > status = "okay"; > }; > > auart1: serial@8006c000 { > pinctrl-names = "default"; > pinctrl-0 = <&auart1_2pins_a>; > status = "okay"; > }; > > auart2: serial@8006e000 { > compatible = "fsl,imx28-ekuart"; > pinctrl-names = "default"; > pinctrl-0 = <&auart2_2pins_a > &rs485_tx_en>; > rs485-rts-active-low; > rts-gpio = <&gpio2 19 1>; > linux,rs485-enabled-at-boot-time; > rs485-rts-delay = <0 0>; > status = "okay"; > }; > > usbphy0: usbphy@8007c000 { > status = "okay"; > }; > > usbphy1: usbphy@8007e000 { > status = "okay"; > }; > }; > }; > > ahb@80080000 { > usb0: usb@80080000 { > pinctrl-names = "default"; > pinctrl-0 = <&usb0_id_pins_b > &usbphy0_pins_a>; > vbus-supply = <&reg_usb0_vbus>; > dr_mode = "otg"; > status = "okay"; > }; > > usb1: usb@80090000 { > pinctrl-names = "default"; > pinctrl-0 = <&usbphy1_pins_a>; > vbus-supply = <&reg_usb1_vbus>; > dr_mode = "host"; > status = "okay"; > }; > > mac0: ethernet@800f0000 { > phy-mode = "rmii"; > pinctrl-names = "default"; > pinctrl-0 = <&mac0_pins_a>; > phy-supply = <&reg_3p3v>; > phy-reset-gpios = <&gpio3 1 0>; > phy-reset-duration = <100>; > status = "okay"; > }; > }; > > regulators { > compatible = "simple-bus"; > > reg_3p3v: 3p3v { > compatible = "regulator-fixed"; > regulator-name = "3P3V"; > regulator-min-microvolt = <3300000>; > regulator-max-microvolt = <3300000>; > regulator-always-on; > }; > > reg_vddio_sd1: vddio-sd1 { > compatible = "regulator-fixed"; > regulator-name = "vddio-sd1"; > regulator-min-microvolt = <3300000>; > regulator-max-microvolt = <3300000>; > gpio = <&gpio3 28 0>; > }; > > reg_usb0_vbus: usb0_vbus { > compatible = "regulator-fixed"; > regulator-name = "usb0_vbus"; > regulator-min-microvolt = <5000000>; > regulator-max-microvolt = <5000000>; > gpio = <&gpio0 4 0>; > enable-active-low; > }; > > reg_usb1_vbus: usb1_vbus { > compatible = "regulator-fixed"; > regulator-name = "usb1_vbus"; > regulator-min-microvolt = <5000000>; > regulator-max-microvolt = <5000000>; > gpio = <&gpio0 18 0>; > enable-active-low; > }; > > reg_lcd_3v3: lcd-3v3 { > compatible = "regulator-fixed"; > regulator-name = "lcd-3v3"; > regulator-min-microvolt = <3300000>; > regulator-max-microvolt = <3300000>; > gpio = <&gpio0 5 0>; > enable-active-high; > }; > }; > > sound { > compatible = "fsl,imx28-evk-sgtl5000", > "fsl,mxs-audio-sgtl5000"; > model = "imx28-evk-sgtl5000"; > saif-controllers = <&saif0 &saif1>; > audio-codec = <&sgtl5000>; > }; > > backlight { > compatible = "pwm-backlight"; > pwms = <&pwm 4 5000000>; > brightness-levels = <0 4 8 16 32 64 128 255>; > default-brightness-level = <6>; > }; > > }; As you can see the audio section _IS_ almost equal. In both boards (IMX28EVK and EK340) the VDDD pin is left floating so it uses the internal ldo voltage regulator. In IMX28EVK and in EK340 the VDDIO is from the VDDIO_3V3 from the SoC pin. In IMX28EVK the VDDA in tied/coupled with a choke (L2 470Ohm) to VDDIO, but in EK340 is using an external 3V3 voltage regulator that is triggered by VDDIO_3V3 to obtain the 3V3 voltage to the VDDA, so I can assume it is _almost_ the same. The same voltage regulator is used as power supply to all i2c stuff (RTC, EEPROM, GPIO Expander,...). VDDIO_3V3 from Soc is quite limited as current output so I choose to use an external 3V3 regulator. If I run: > arecord -t wav -d 5 -d hw:0,1 -f S16_LE -c 2 /tmp/sample.wav in the IMX28EVK it works as expected (I can clearly see I2C lines moving and SAIF1 lines moving too...) in the EK340 I can see the same lines moving (I2C and SAIF1) _BUT_ it hangs for 10 seconds and fails with an error: > arecord: pcm_read:1801: read error: Input/output error and the only thing it writes it is the WAV HEADER (44 bytes). I saw all stuff regarding SGTL5000 issues on the FSL Community: > https://community.freescale.com/thread/300291 > https://community.freescale.com/message/328708#328708 > https://community.freescale.com/thread/307611#331003 and this morning I found this too: > https://community.freescale.com/message/496426#496426 Now I can check this out too, but I am very UNCONFIDENT it will NO USEFUL to me, as in the EVK everything is working, meanwhile in EK340 it does not work for record, but only for playback... :-( Last but not least the boards are different by SoC version: IMX28EVK is using i.MX287 EK340 is using i.MX286 Any help, clue or whatever I can check and try to find out this problem??? Thank you in advance, Gianluca Renzi -- Eurek s.r.l. | Electronic Engineering | http://www.eurek.it via Celletta 8/B, 40026 Imola, Italy | Phone: +39-(0)542-609120 p.iva 00690621206 - c.f. 04020030377 | Fax: +39-(0)542-609212

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[alsa-devel] [asoc:topic/zx 2/4] sound/soc/zte/zx296702-i2s.c:358:27: sparse: symbol 'zx_i2s_dai' was not declared. Should it be static?
by kbuild test robot 04 Jun '15

04 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/zx head: dc772a4cf76113d7269e4fb1c45e5d85c0cf458e commit: e5d4cd87800ce12c356e7eb571e565e839ab3a90 [2/4] ASoC: zx: Add ZTE zx296702 I2S DAI driver reproduce: # apt-get install sparse git checkout e5d4cd87800ce12c356e7eb571e565e839ab3a90 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/zte/zx296702-i2s.c:358:27: sparse: symbol 'zx_i2s_dai' was not declared. Should it be static? Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

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[alsa-devel] [PATCH 00/22] ASoC: tas2552: Fixes, cleanups and improvements
by Peter Ujfalusi 04 Jun '15

04 Jun '15

Hi, 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 (22): 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: 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] [PATCH 0/7] ASoC: remove bitwise operations on GPIO level value
by Vladimir Zapolskiy 04 Jun '15

04 Jun '15

The series does not contain any functional changes, it touches only implementation of gpiolib .set and .direction_output callbacks. The main intention of the change is to remove bitwise operations on GPIO high/low level value as a preceding change before updating gpiolib callback signatures to utilize bool type as a representation of GPIO level. The change covers all input cases of GPIO level (i.e. .set and .direction_output) in sound/*, also the series contains a small clean-ups in rt5677 and wm8903 codec drivers related to gpiolib callbacks. Vladimir Zapolskiy (7): ASoC: rt5677: add GPIO helper macros ASoC: rt5677: clean up gpiolib callbacks ASoC: wm8903: generalize GPIO control register bits ASoC: wm8903: simplify gpiolib callbacks ASoC: wm5100: remove bitwise operations involving GPIO level value ASoC: wm8962: remove bitwise operations involving GPIO level value ASoC: wm8996: remove bitwise operations involving GPIO level value include/sound/wm8903.h | 222 ++++++++-------------------------------------- sound/soc/codecs/rt5677.c | 32 +++++-- sound/soc/codecs/rt5677.h | 6 ++ sound/soc/codecs/wm5100.c | 21 ++--- sound/soc/codecs/wm8903.c | 44 ++++----- sound/soc/codecs/wm8962.c | 13 ++- sound/soc/codecs/wm8996.c | 11 ++- 7 files changed, 112 insertions(+), 237 deletions(-) -- 2.1.4

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[alsa-devel] [asoc:topic/topology 104/105] sound/soc/soc-topology.c:1672:45: sparse: restricted __le32 degrades to integer
by kbuild test robot 04 Jun '15

04 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/topology head: 932ae8809469770a07ce19d6967d2ce303befa08 commit: 8a9782346dccd82cf912552735bda619de4efd8c [104/105] ASoC: topology: Add topology core reproduce: # apt-get install sparse git checkout 8a9782346dccd82cf912552735bda619de4efd8c make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/soc-topology.c:1672:45: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:189:28: sparse: restricted __le32 degrades to integer >> sound/soc/soc-topology.c:190:39: sparse: incorrect type in return expression (different base types) sound/soc/soc-topology.c:190:39: expected int sound/soc/soc-topology.c:190:39: got restricted __le32 [usertype] reg sound/soc/soc-topology.c:202:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:203:39: sparse: incorrect type in return expression (different base types) sound/soc/soc-topology.c:203:39: expected int sound/soc/soc-topology.c:203:39: got restricted __le32 [usertype] shift sound/soc/soc-topology.c:227:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:227:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:227:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:227:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:227:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:227:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:227:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:227:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:247:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:247:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:247:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:247:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:247:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:247:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:247:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:518:42: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:520:42: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:522:42: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:533:61: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:535:61: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:537:61: sparse: restricted __le32 degrades to integer >> sound/soc/soc-topology.c:601:20: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:601:20: expected unsigned int [unsigned] numid sound/soc/soc-topology.c:601:20: got restricted __le32 [usertype] numid >> sound/soc/soc-topology.c:602:21: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:602:21: expected unsigned int [unsigned] length sound/soc/soc-topology.c:602:21: got restricted __le32 [usertype] size >> sound/soc/soc-topology.c:603:9: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:603:9: expected unsigned long [unsigned] [usertype] len sound/soc/soc-topology.c:603:9: got restricted __le32 [usertype] size sound/soc/soc-topology.c:644:33: sparse: restricted __le32 degrades to integer >> sound/soc/soc-topology.c:654:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:654:27: expected unsigned int [unsigned] [addressable] [assigned] access sound/soc/soc-topology.c:654:27: got restricted __le32 [usertype] access >> sound/soc/soc-topology.c:656:26: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:656:26: expected int [signed] max sound/soc/soc-topology.c:656:26: got restricted __le32 [usertype] max sound/soc/soc-topology.c:725:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:735:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:735:27: expected unsigned int [unsigned] [addressable] [assigned] access sound/soc/soc-topology.c:735:27: got restricted __le32 [usertype] access sound/soc/soc-topology.c:747:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:747:25: expected int [signed] max sound/soc/soc-topology.c:747:25: got restricted __le32 [usertype] max >> sound/soc/soc-topology.c:748:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:748:25: expected int [signed] min sound/soc/soc-topology.c:748:25: got restricted __le32 [usertype] min >> sound/soc/soc-topology.c:749:28: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:749:28: expected unsigned int [unsigned] invert:1 sound/soc/soc-topology.c:749:28: got restricted __le32 [usertype] invert >> sound/soc/soc-topology.c:750:34: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:750:34: expected int [signed] platform_max sound/soc/soc-topology.c:750:34: got restricted __le32 [usertype] platform_max >> sound/soc/soc-topology.c:776:55: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:776:55: expected unsigned int [unsigned] [usertype] tlv_size sound/soc/soc-topology.c:776:55: got restricted __le32 [usertype] tlv_size sound/soc/soc-topology.c:801:44: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:805:27: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:832:15: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:836:27: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:840:9: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:864:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:882:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:882:27: expected unsigned int [unsigned] [addressable] [assigned] access sound/soc/soc-topology.c:882:27: got restricted __le32 [usertype] access >> sound/soc/soc-topology.c:890:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:890:27: expected unsigned int [unsigned] items sound/soc/soc-topology.c:890:27: got restricted __le32 [usertype] items >> sound/soc/soc-topology.c:891:26: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:891:26: expected unsigned int [unsigned] mask sound/soc/soc-topology.c:891:26: got restricted __le32 [usertype] mask sound/soc/soc-topology.c:897:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:897:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:897:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:897:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:897:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:971:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:971:45: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:978:28: sparse: restricted __le32 degrades to integer >> sound/soc/soc-topology.c:990:60: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:990:60: expected unsigned long [unsigned] [usertype] size sound/soc/soc-topology.c:990:60: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:997:59: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:997:59: expected unsigned long [unsigned] [usertype] size sound/soc/soc-topology.c:997:59: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:1000:60: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:1000:60: expected unsigned long [unsigned] [usertype] size sound/soc/soc-topology.c:1000:60: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:982:41: sparse: restricted __le32 degrades to integer >> sound/soc/soc-topology.c:1017:24: sparse: incorrect type in initializer (different base types) sound/soc/soc-topology.c:1017:24: expected int [signed] count sound/soc/soc-topology.c:1017:24: got restricted __le32 [usertype] count sound/soc/soc-topology.c:1020:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1020:45: sparse: restricted __le32 degrades to integer >> sound/soc/soc-topology.c:1026:27: sparse: incorrect type in argument 4 (different base types) sound/soc/soc-topology.c:1026:27: expected unsigned long [unsigned] [usertype] bytes sound/soc/soc-topology.c:1026:27: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:1084:33: sparse: restricted __le32 degrades to integer >> sound/soc/soc-topology.c:1097:30: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1097:30: expected unsigned int [unsigned] access sound/soc/soc-topology.c:1097:30: got restricted __le32 [usertype] access sound/soc/soc-topology.c:1109:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1109:25: expected int [signed] max sound/soc/soc-topology.c:1109:25: got restricted __le32 [usertype] max sound/soc/soc-topology.c:1110:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1110:25: expected int [signed] min sound/soc/soc-topology.c:1110:25: got restricted __le32 [usertype] min sound/soc/soc-topology.c:1111:28: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1111:28: expected unsigned int [unsigned] invert:1 sound/soc/soc-topology.c:1111:28: got restricted __le32 [usertype] invert sound/soc/soc-topology.c:1112:34: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1112:34: expected int [signed] platform_max sound/soc/soc-topology.c:1112:34: got restricted __le32 [usertype] platform_max sound/soc/soc-topology.c:1156:25: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1177:20: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1177:20: expected unsigned int [unsigned] access sound/soc/soc-topology.c:1177:20: got restricted __le32 [usertype] access sound/soc/soc-topology.c:1184:19: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1184:19: expected unsigned int [unsigned] items sound/soc/soc-topology.c:1184:19: got restricted __le32 [usertype] items sound/soc/soc-topology.c:1185:18: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1185:18: expected unsigned int [unsigned] mask sound/soc/soc-topology.c:1185:18: got restricted __le32 [usertype] mask sound/soc/soc-topology.c:1188:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1188:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1188:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1188:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1188:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1236:27: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1271:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1281:30: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1281:30: expected unsigned int [unsigned] access sound/soc/soc-topology.c:1281:30: got restricted __le32 [usertype] access sound/soc/soc-topology.c:1283:26: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1283:26: expected int [signed] max sound/soc/soc-topology.c:1283:26: got restricted __le32 [usertype] max >> sound/soc/soc-topology.c:1339:38: sparse: incorrect type in argument 1 (different base types) sound/soc/soc-topology.c:1339:38: expected int [signed] tplg_type sound/soc/soc-topology.c:1339:38: got restricted __le32 [usertype] id >> sound/soc/soc-topology.c:1351:22: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1351:22: expected int [signed] [addressable] [assigned] reg sound/soc/soc-topology.c:1351:22: got restricted __le32 [usertype] reg >> sound/soc/soc-topology.c:1352:24: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1352:24: expected unsigned char [unsigned] [addressable] [assigned] shift sound/soc/soc-topology.c:1352:24: got restricted __le32 [usertype] shift >> sound/soc/soc-topology.c:1353:23: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1353:23: expected unsigned int [unsigned] [addressable] [assigned] mask sound/soc/soc-topology.c:1353:23: got restricted __le32 [usertype] mask >> sound/soc/soc-topology.c:1361:67: sparse: too many warnings vim +1672 sound/soc/soc-topology.c 1656 1657 while (!soc_tplg_is_eof(tplg)) { 1658 1659 /* make sure header is valid before loading */ 1660 ret = soc_valid_header(tplg, hdr); 1661 if (ret < 0) 1662 return ret; 1663 else if (ret == 0) 1664 break; 1665 1666 /* load the header object */ 1667 ret = soc_tplg_load_header(tplg, hdr); 1668 if (ret < 0) 1669 return ret; 1670 1671 /* goto next header */ > 1672 tplg->hdr_pos += hdr->payload_size + 1673 sizeof(struct snd_soc_tplg_hdr); 1674 hdr = (struct snd_soc_tplg_hdr *)tplg->hdr_pos; 1675 } 1676 1677 /* next data type pass */ 1678 tplg->pass++; 1679 } 1680 --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

1 0

[alsa-devel] [PATCH v3 0/6] Support ZTE zx296702 SPDIF/I2S
by Jun Nie 04 Jun '15

04 Jun '15

Changes vs v2: - remove pcm dma driver and merge the related call into DAI driver - Add zx HDMI codec with direct HDMI function call - Other typo and minor fix Jun Nie (6): dt: Add documentation for the ZTE SPDIF controller dt: Add documentation for the ZTE I2S controller dt: Add doc for the ZTE HDMI audio codec binding ASoC: zx: Add zx296702 SPDIF support ASoC: zx: Add ZTE zx296702 I2S DAI driver ASoC: zx: add zx296702 hdmi codec .../devicetree/bindings/sound/zte,zx-hdmi.txt | 10 + .../devicetree/bindings/sound/zte,zx-i2s.txt | 44 +++ .../devicetree/bindings/sound/zte,zx-spdif.txt | 28 ++ include/sound/zx_hdmi_audio.h | 7 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/zx296702_hdmi.c | 121 ++++++ sound/soc/zte/Kconfig | 17 + sound/soc/zte/Makefile | 2 + sound/soc/zte/zx296702-i2s.c | 437 +++++++++++++++++++++ sound/soc/zte/zx296702-spdif.c | 370 +++++++++++++++++ 13 files changed, 1042 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-spdif.txt create mode 100644 include/sound/zx_hdmi_audio.h create mode 100644 sound/soc/codecs/zx296702_hdmi.c create mode 100644 sound/soc/zte/Kconfig create mode 100644 sound/soc/zte/Makefile create mode 100644 sound/soc/zte/zx296702-i2s.c create mode 100644 sound/soc/zte/zx296702-spdif.c -- 1.9.1

3 18

Re: [alsa-devel] [PATCH] Added support for Creative SoundBlaster Omni Surround 5.1 USB sound card
by Raymond Yau 04 Jun '15

04 Jun '15

>> >> Have you tested surround21? > > Yes, front speakers work fine, however LFE doesn't seem to work. Instead when I try to test LFE (from sound UI) I have weird noise in one of rear speakers. There are no such problems with any other mode though (I see that officially only 5.1 mode is supported, however 4.1, 5.0 and 5.1 all work fine), I decided to leave 2.1 mode for now, not sure whose fault is that. I: [pulseaudio] (alsa-lib)pcm_hw.c: Invalid PCM state for chmap_get: OPEN I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-5) http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=test/chmap.c;hb=HEAD Do snd-usb-audio provide channel map ? http://cgit.freedesktop.org/pulseaudio/pulseaudio/log/?qt=grep&q=lfe-filter http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/USB-Aud… # If a device does not use the first PCM device for digital data, the device # number for the iec958 device can be changed here. USB-Audio.pcm.iec958_device Do the red light in the optical spdif alway turn on ? Do the red light turn on when pulseaudio probe the devices ? card 2: S51 [SB Omni Surround 5.1], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: S51 [SB Omni Surround 5.1], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: S51 [SB Omni Surround 5.1], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0

1 0

[alsa-devel] [PATCH v3 1/3] ASoC: topology: Add topology UAPI header
by Liam Girdwood 03 Jun '15

03 Jun '15

The ASoC topology UAPI header defines the structures required to define any DSP firmware audio topology and control objects from userspace. The following objects are supported :- o kcontrols including TLV controls. o DAPM widgets and graph elements o Vendor bespoke objects. o Coefficient data o FE PCM capabilities and config. o BE link capabilities and config. o Codec <-> codec link capabilities and config. o Topology object manifest. The file format is simple and divided into blocks for each object type and each block has a header that defines it's size and type. Blocks can be in any order of type and can either all be in a single file or spread across more than one file. Blocks also have a group identifier ID so that they can be loaded and unloaded by ID. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dapm.h | 1 + include/uapi/sound/asoc.h | 388 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 389 insertions(+) create mode 100644 include/uapi/sound/asoc.h diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 1065095..6f440b2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -15,6 +15,7 @@ #include <linux/types.h> #include <sound/control.h> +#include <sound/asoc.h> struct device; diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h new file mode 100644 index 0000000..1221520 --- /dev/null +++ b/include/uapi/sound/asoc.h @@ -0,0 +1,388 @@ +/* + * uapi/sound/asoc.h -- ALSA SoC Firmware Controls and DAPM + * + * Copyright (C) 2012 Texas Instruments Inc. + * Copyright (C) 2015 Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, + * algorithms, equalisers, DAIs, widgets etc. +*/ + +#ifndef __LINUX_UAPI_SND_ASOC_H +#define __LINUX_UAPI_SND_ASOC_H + +#include <linux/types.h> +#include <sound/asound.h> + +/* + * Maximum number of channels topology kcontrol can represent. + */ +#define SND_SOC_TPLG_MAX_CHAN 8 + +/* + * Maximum number of PCM formats capability + */ +#define SND_SOC_TPLG_MAX_FORMATS 16 + +/* + * Maximum number of PCM stream configs + */ +#define SND_SOC_TPLG_STREAM_CONFIG_MAX 8 + +/* individual kcontrol info types - can be mixed with other types */ +#define SND_SOC_TPLG_CTL_VOLSW 1 +#define SND_SOC_TPLG_CTL_VOLSW_SX 2 +#define SND_SOC_TPLG_CTL_VOLSW_XR_SX 3 +#define SND_SOC_TPLG_CTL_ENUM 4 +#define SND_SOC_TPLG_CTL_BYTES 5 +#define SND_SOC_TPLG_CTL_ENUM_VALUE 6 +#define SND_SOC_TPLG_CTL_RANGE 7 +#define SND_SOC_TPLG_CTL_STROBE 8 + + +/* individual widget kcontrol info types - can be mixed with other types */ +#define SND_SOC_TPLG_DAPM_CTL_VOLSW 64 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE 65 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT 66 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE 67 +#define SND_SOC_TPLG_DAPM_CTL_PIN 68 + +/* DAPM widget types - add new items to the end */ +#define SND_SOC_TPLG_DAPM_INPUT 0 +#define SND_SOC_TPLG_DAPM_OUTPUT 1 +#define SND_SOC_TPLG_DAPM_MUX 2 +#define SND_SOC_TPLG_DAPM_MIXER 3 +#define SND_SOC_TPLG_DAPM_PGA 4 +#define SND_SOC_TPLG_DAPM_OUT_DRV 5 +#define SND_SOC_TPLG_DAPM_ADC 6 +#define SND_SOC_TPLG_DAPM_DAC 7 +#define SND_SOC_TPLG_DAPM_SWITCH 8 +#define SND_SOC_TPLG_DAPM_PRE 9 +#define SND_SOC_TPLG_DAPM_POST 10 +#define SND_SOC_TPLG_DAPM_AIF_IN 11 +#define SND_SOC_TPLG_DAPM_AIF_OUT 12 +#define SND_SOC_TPLG_DAPM_DAI_IN 13 +#define SND_SOC_TPLG_DAPM_DAI_OUT 14 +#define SND_SOC_TPLG_DAPM_DAI_LINK 15 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK + +/* Header magic number and string sizes */ +#define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ + +/* string sizes */ +#define SND_SOC_TPLG_NUM_TEXTS 16 + +/* ABI version */ +#define SND_SOC_TPLG_ABI_VERSION 0x2 + +/* Max size of TLV data */ +#define SND_SOC_TPLG_TLV_SIZE 32 + +/* + * File and Block header data types. + * Add new generic and vendor types to end of list. + * Generic types are handled by the core whilst vendors types are passed + * to the component drivers for handling. + */ +#define SND_SOC_TPLG_TYPE_MIXER 1 +#define SND_SOC_TPLG_TYPE_BYTES 2 +#define SND_SOC_TPLG_TYPE_ENUM 3 +#define SND_SOC_TPLG_TYPE_DAPM_GRAPH 4 +#define SND_SOC_TPLG_TYPE_DAPM_WIDGET 5 +#define SND_SOC_TPLG_TYPE_DAI_LINK 6 +#define SND_SOC_TPLG_TYPE_PCM 7 +#define SND_SOC_TPLG_TYPE_MANIFEST 8 +#define SND_SOC_TPLG_TYPE_CODEC_LINK 9 +#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_CODEC_LINK + +/* vendor block IDs - please add new vendor types to end */ +#define SND_SOC_TPLG_TYPE_VENDOR_FW 1000 +#define SND_SOC_TPLG_TYPE_VENDOR_CONFIG 1001 +#define SND_SOC_TPLG_TYPE_VENDOR_COEFF 1002 +#define SND_SOC_TPLG_TYPEVENDOR_CODEC 1003 + +#define SND_SOC_TPLG_STREAM_PLAYBACK 0 +#define SND_SOC_TPLG_STREAM_CAPTURE 1 + +/* + * Block Header. + * This header preceeds all object and object arrays below. + */ +struct snd_soc_tplg_hdr { + __le32 magic; /* magic number */ + __le32 abi; /* ABI version */ + __le32 version; /* optional vendor specific version details */ + __le32 type; /* SND_SOC_TPLG_TYPE_ */ + __le32 size; /* size of this structure */ + __le32 vendor_type; /* optional vendor specific type info */ + __le32 payload_size; /* data bytes, excluding this header */ + __le32 index; /* identifier for block */ + __le32 count; /* number of elements in block */ +} __attribute__((packed)); + +/* + * Private data. + * All topology objects may have private data that can be used by the driver or + * firmware. Core will ignore this data. + */ +struct snd_soc_tplg_private { + __le32 size; /* in bytes of private data */ + char data[0]; +} __attribute__((packed)); + +/* + * Kcontrol TLV data. + */ +struct snd_soc_tplg_ctl_tlv { + __le32 size; /* in bytes aligned to 4 */ + __le32 numid; /* control element numeric identification */ + __le32 count; /* number of elem in data array */ + __le32 data[SND_SOC_TPLG_TLV_SIZE]; +} __attribute__((packed)); + +/* + * Kcontrol channel data + */ +struct snd_soc_tplg_channel { + __le32 size; /* in bytes of this structure */ + __le32 reg; + __le32 shift; + __le32 id; /* ID maps to Left, Right, LFE etc */ +} __attribute__((packed)); + +/* + * Kcontrol Operations IDs + */ +struct snd_soc_tplg_kcontrol_ops_id { + __le32 get; + __le32 put; + __le32 info; +} __attribute__((packed)); + +/* + * kcontrol header + */ +struct snd_soc_tplg_ctl_hdr { + __le32 size; /* in bytes of this structure */ + __le32 type; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 access; + struct snd_soc_tplg_kcontrol_ops_id ops; + __le32 tlv_size; /* non zero means control has TLV data */ +} __attribute__((packed)); + +/* + * Stream Capabilities + */ +struct snd_soc_tplg_stream_caps { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le64 formats[SND_SOC_TPLG_MAX_FORMATS]; /* supported formats SNDRV_PCM_FMTBIT_* */ + __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ + __le32 rate_min; /* min rate */ + __le32 rate_max; /* max rate */ + __le32 channels_min; /* min channels */ + __le32 channels_max; /* max channels */ + __le32 periods_min; /* min number of periods */ + __le32 periods_max; /* max number of periods */ + __le32 period_size_min; /* min period size bytes */ + __le32 period_size_max; /* max period size bytes */ + __le32 buffer_size_min; /* min buffer size bytes */ + __le32 buffer_size_max; /* max buffer size bytes */ +} __attribute__((packed)); + +/* + * FE or BE Stream configuration supported by SW/FW + */ +struct snd_soc_tplg_stream { + __le32 size; /* in bytes of this structure */ + __le64 format; /* SNDRV_PCM_FMTBIT_* */ + __le32 rate; /* SNDRV_PCM_RATE_* */ + __le32 period_bytes; /* size of period in bytes */ + __le32 buffer_bytes; /* size of buffer in bytes */ + __le32 channels; /* channels */ + __le32 tdm_slot; /* optional BE bitmask of supported TDM slots */ + __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ +} __attribute__((packed)); + +/* + * Duplex stream configuration supported by SW/FW. + */ +struct snd_soc_tplg_stream_config { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct snd_soc_tplg_stream playback; + struct snd_soc_tplg_stream capture; +} __attribute__((packed)); + +/* + * Manifest. List totals for each payload type. Not used in parsing, but will + * be passed to the component driver before any other objects in order for any + * global componnent resource allocations. + * + * File block representation for manifest :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_manifest | 1 | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_manifest { + __le32 size; /* in bytes of this structure */ + __le32 control_elems; /* number of control elements */ + __le32 widget_elems; /* number of widget elements */ + __le32 graph_elems; /* number of graph elements */ + __le32 dai_elems; /* number of DAI elements */ + __le32 dai_link_elems; /* number of DAI link elements */ +} __attribute__((packed)); + +/* + * Mixer kcontrol. + * + * File block representation for mixer kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_mixer_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_mixer_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 min; + __le32 max; + __le32 platform_max; + __le32 invert; + __le32 num_channels; + struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; + struct snd_soc_tplg_ctl_tlv tlv; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Enumerated kcontrol + * + * File block representation for enum kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_enum_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_enum_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 num_channels; + struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; + __le32 items; + __le32 mask; + __le32 count; + char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4]; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Bytes kcontrol + * + * File block representation for bytes kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_bytes_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_bytes_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 max; + __le32 mask; + __le32 base; + __le32 num_regs; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * DAPM Graph Element + * + * File block representation for DAPM graph elements :- + * +-------------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-------------------------------------+----+ + * | struct snd_soc_tplg_dapm_graph_elem | N | + * +-------------------------------------+----+ + */ +struct snd_soc_tplg_dapm_graph_elem { + char sink[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char control[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char source[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; +} __attribute__((packed)); + +/* + * DAPM Widget. + * + * File block representation for DAPM widget :- + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_dapm_widget | N | + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_enum_control | 0|1 | + * | struct snd_soc_tplg_mixer_control | 0|N | + * +-------------------------------------+-----+ + * + * Optional enum or mixer control can be appended to the end of each widget + * in the block. + */ +struct snd_soc_tplg_dapm_widget { + __le32 size; /* in bytes of this structure */ + __le32 id; /* SND_SOC_DAPM_CTL */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char sname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + __le32 reg; /* negative reg = no direct dapm */ + __le32 shift; /* bits to shift */ + __le32 mask; /* non-shifted mask */ + __u32 invert; /* invert the power bit */ + __u32 ignore_suspend; /* kept enabled over suspend */ + __u16 event_flags; + __u16 event_type; + __u16 num_kcontrols; + struct snd_soc_tplg_private priv; + /* + * kcontrols that relate to this widget + * follow here after widget private data + */ +} __attribute__((packed)); + +struct snd_soc_tplg_pcm_cfg_caps { + struct snd_soc_tplg_stream_caps caps; + struct snd_soc_tplg_stream_config configs[SND_SOC_TPLG_STREAM_CONFIG_MAX]; + __le32 num_configs; /* number of configs */ +} __attribute__((packed)); + +/* + * Describes SW/FW specific features of PCM or DAI link. + * + * File block representation for PCM/DAI-Link :- + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_dapm_pcm_dai | N | + * +-----------------------------------+-----+ + */ +struct snd_soc_tplg_pcm_dai { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 id; /* unique ID - used to match */ + __le32 playback; /* supports playback mode */ + __le32 capture; /* supports capture mode */ + __le32 compress; /* 1 = compressed; 0 = PCM */ + struct snd_soc_tplg_pcm_cfg_caps capconf[2]; /* capabilities and configs */ +} __attribute__((packed)); + +#endif -- 2.1.4

3 5

[alsa-devel] [PATCH] ASoC: fsl: Correct FSL_SAI_CR2_MSEL_MASK definition
by Takashi Iwai 03 Jun '15

03 Jun '15

The bit mask FSL_SAI_CR2_MSEL_MASK is defined as (0xff << 26) and this obviously overflows the 32bit integer. Also this clears other bits than the needed ones (FSL_SAI_CR2_MSEL_XXX) wrongly. This patch fixes the definition only to mask bits 26 and 27. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=84351 Reported-by: David Binderman <dcb314(a)hotmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/fsl/fsl_sai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 34667209b607..0e33a17ddd01 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -72,7 +72,7 @@ /* SAI Transmit and Recieve Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) -#define FSL_SAI_CR2_MSEL_MASK (0xff << 26) +#define FSL_SAI_CR2_MSEL_MASK (0x03 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 #define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) #define FSL_SAI_CR2_MSEL_MCLK2 BIT(27) -- 2.4.2

2 2

[alsa-devel] simple-audio-card with omap4-dmic
by Jack Mitchell 03 Jun '15

03 Jun '15

I'm currently working with a PandaBoard clone (OMAP44xx) which doesn't have the usual twl6040 codec, however I would like to attach a digital microphone directly into the OMAP. I have added the following to my device tree: sound { compatible = "simple-audio-card"; simple-audio-card,name = "dmic-card"; simple-audio-card,format = "pdm"; status = "okay"; simple-audio-card,dai-link@0 { cpu { sound-dai = <&dmic>; }; codec { sound-dai = <&dmic_codec>; }; }; }; dmic_codec: dmic_codec { #sound-dai-cells = <0>; compatible = "linux,dmic-codec"; }; and created a small patch to allow the generic dmic-hifi codec to be instantiated from device tree. diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index fde5325..67b9198 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -69,11 +69,18 @@ static int dmic_dev_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:dmic-codec"); +#ifdef CONFIG_OF +static const struct of_device_id dmic_dt_ids[] = { + { .compatible = "linux,dmic-codec", }, + { } +}; +MODULE_DEVICE_TABLE(of, dmic_dt_ids); +#endif static struct platform_driver dmic_driver = { .driver = { .name = "dmic-codec", + .of_match_table = of_match_ptr(dmic_dt_ids), }, .probe = dmic_dev_probe, .remove = dmic_dev_remove, @@ -81,6 +88,7 @@ static struct platform_driver dmic_driver = { module_platform_driver(dmic_driver); +MODULE_ALIAS("platform:dmic-codec"); MODULE_DESCRIPTION("Generic DMIC driver"); MODULE_AUTHOR("Liam Girdwood <lrg(a)slimlogic.co.uk>"); MODULE_LICENSE("GPL"); However, no soundcards get loaded, and I don't get error or warnings in dmesg, so I'm at a bit of a loss at where I'm going wrong. Can anybody see any glaring mistakes? I know the driver is definitely getting loaded as I have seen errors relating to me not specifying the sound-dai-cells from the simple-audio-card driver. Any ideas would be greatly recieved! Cheers,

1 1

[alsa-devel] [PATCH v2 1/2] ASoC: rt5645: change gpio to gpiod APIs
by Bard Liao 03 Jun '15

03 Jun '15

Move gpio to gpio_desc and use gpiod APIs in codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt5645.h | 3 --- sound/soc/codecs/rt5645.c | 47 +++++++++++++---------------------------------- sound/soc/codecs/rt5645.h | 1 + 3 files changed, 14 insertions(+), 37 deletions(-) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index 652cb9e..22734bc 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -20,9 +20,6 @@ struct rt5645_platform_data { unsigned int dmic2_data_pin; /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */ - unsigned int hp_det_gpio; - bool gpio_hp_det_active_high; - unsigned int jd_mode; }; diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index aaede45..0cb4942 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2938,17 +2938,11 @@ static int rt5645_irq_detection(struct rt5645_priv *rt5645) switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ - if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) { - gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); - dev_dbg(rt5645->codec->dev, "gpio = %d(%d)\n", - rt5645->pdata.hp_det_gpio, gpio_state); - } - if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || - (!rt5645->pdata.gpio_hp_det_active_high && - !gpio_state)) { - report = rt5645_jack_detect(rt5645->codec, 1); - } else { - report = rt5645_jack_detect(rt5645->codec, 0); + if (rt5645->gpiod_hp_det) { + gpio_state = gpiod_get_value(rt5645->gpiod_hp_det); + dev_dbg(rt5645->codec->dev, "gpio_state = %d\n", + gpio_state); + report = rt5645_jack_detect(rt5645->codec, gpio_state); } snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); @@ -3253,19 +3247,17 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } else { if (dmi_check_system(dmi_platform_intel_braswell)) { rt5645->pdata = *rt5645_pdata; - gpiod = devm_gpiod_get_index(&i2c->dev, "rt5645", 0); - - if (IS_ERR(gpiod) || gpiod_direction_input(gpiod)) { - rt5645->pdata.hp_det_gpio = -1; - dev_err(&i2c->dev, "failed to initialize gpiod\n"); - } else { - rt5645->pdata.hp_det_gpio = desc_to_gpio(gpiod); - rt5645->pdata.gpio_hp_det_active_high - = !gpiod_is_active_low(gpiod); - } } } + rt5645->gpiod_hp_det = devm_gpiod_get_index(&i2c->dev, "rt5645", 0); + + if (IS_ERR(rt5645->gpiod_hp_det) || + gpiod_direction_input(rt5645->gpiod_hp_det)) { + rt5645->gpiod_hp_det = NULL; + dev_err(&i2c->dev, "failed to initialize gpiod\n"); + } + rt5645->regmap = devm_regmap_init_i2c(i2c, &rt5645_regmap); if (IS_ERR(rt5645->regmap)) { ret = PTR_ERR(rt5645->regmap); @@ -3425,16 +3417,6 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); } - if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) { - ret = gpio_request(rt5645->pdata.hp_det_gpio, "rt5645"); - if (ret) - dev_err(&i2c->dev, "Fail gpio_request hp_det_gpio\n"); - - ret = gpio_direction_input(rt5645->pdata.hp_det_gpio); - if (ret) - 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, @@ -3450,9 +3432,6 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) cancel_delayed_work_sync(&rt5645->jack_detect_work); - if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) - gpio_free(rt5645->pdata.hp_det_gpio); - snd_soc_unregister_codec(&i2c->dev); return 0; diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 9ec4e89..0353a6a 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2182,6 +2182,7 @@ struct rt5645_priv { struct rt5645_platform_data pdata; struct regmap *regmap; struct i2c_client *i2c; + struct gpio_desc *gpiod_hp_det; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; struct snd_soc_jack *btn_jack; -- 1.8.1.1.439.g50a6b54

4 8

[alsa-devel] [PATCH] ALSA: usb-audio: fix missing input volume controls in MAYA44 USB(+)
by Clemens Ladisch 03 Jun '15

03 Jun '15

The driver worked around an error in the MAYA44 USB(+)'s mixer unit descriptor by aborting before parsing the missing field. However, aborting parsing too early prevented parsing of the other units connected to this unit, so the capture mixer controls would be missing. Fix this by moving the check for this descriptor error after the parsing of the unit's input pins. Reported-by: nightmixes <nightmixes(a)gmail.com> Tested-by: nightmixes <nightmixes(a)gmail.com> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1583,18 +1583,15 @@ static int parse_audio_mixer_unit(struct unitid); return -EINVAL; } - /* no bmControls field (e.g. Maya44) -> ignore */ - if (desc->bLength <= 10 + input_pins) { - usb_audio_dbg(state->chip, "MU %d has no bmControls field\n", - unitid); - return 0; - } num_ins = 0; ich = 0; for (pin = 0; pin < input_pins; pin++) { err = parse_audio_unit(state, desc->baSourceID[pin]); if (err < 0) + continue; + /* no bmControls field (e.g. Maya44) -> ignore */ + if (desc->bLength <= 10 + input_pins) continue; err = check_input_term(state, desc->baSourceID[pin], &iterm); if (err < 0)

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: add MAYA44 USB+ mixer control names
by Clemens Ladisch 03 Jun '15

03 Jun '15

Add mixer control names for the ESI Maya44 USB+ (which appears to be identical width the AudioTrak Maya44 USB). Reported-by: nightmixes <nightmixes(a)gmail.com> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -437,6 +437,11 @@ static struct usbmix_ctl_map usbmix_ctl_ .map = ebox44_map, }, { + /* MAYA44 USB+ */ + .id = USB_ID(0x2573, 0x0008), + .map = maya44_map, + }, + { /* KEF X300A */ .id = USB_ID(0x27ac, 0x1000), .map = scms_usb3318_map,

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[alsa-devel] [PATCH v5 0/2] Add HDA codec match and new registers
by Vinod Koul 03 Jun '15

03 Jun '15

This series is split from SKL driver series and adds default codec match function and then the new HDA registers definitions added in SKL Jeeja KP (1): ALSA: hda - add new HDA registers Vinod Koul (1): ALSA: hda - add HDA default codec match function include/sound/hda_register.h | 88 ++++++++++++++++++++++++++++++++++++++++++++ include/sound/hdaudio.h | 14 +++++++ sound/hda/hda_bus_type.c | 41 +++++++++++++++++++++ 3 files changed, 143 insertions(+) -- 1.9.1

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[alsa-devel] [PATCH 0/5] OMAPDSS: HDMI audio compliance fixes
by Jyri Sarha 03 Jun '15

03 Jun '15

I have rebased these patches from ti-linux-3.14.y on top of Linux 4.0.0. I tested them briefly on OMAP4 and OMAP5 in my environment, but I could not test any channel count beyond stereo. However, there were no conflicts in the rebase and each fix makes sense when looking at the chip TRM and the relevant HDMI specs. All the changes are really minor, but most of them apply to OMAPDSS side, so it is probably best to merge them trough there. Best regards, Jyri Misael Lopez Cruz (5): OMAPDSS: HDMI4: Set correct CC for 8-channels layout OMAPDSS: HDMI5: Set valid sample order OMAPDSS: HDMI5: Fix AUDICONF3 bitfield offsets ASoC: omap-hdmi-audio: Force channel allocation only for OMAP4 ASoC: omap-hdmi-audio: Fix invalid combination of DM_INH and CA drivers/video/fbdev/omap2/dss/hdmi4_core.c | 12 +++++++++++- drivers/video/fbdev/omap2/dss/hdmi5_core.c | 5 ++++- sound/soc/omap/omap-hdmi-audio.c | 12 +++++++----- 3 files changed, 22 insertions(+), 7 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: qcom: Remove QCOM dependency from LPASS variants
by Kenneth Westfield 03 Jun '15

03 Jun '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Building with the following config options ... CONFIG_COMPILE_TEST=y # CONFIG_SND_SOC_QCOM is not set CONFIG_SND_SOC_LPASS_IPQ806X=m CONFIG_SND_SOC_STORM=m ... causes the following build warning: warning: (SND_SOC_STORM) selects SND_SOC_LPASS_IPQ806X which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && SND_SOC_QCOM) To fix this, the dependency on SND_SOC_QCOM from the user-invisible LPASS variant config options has been removed. This will allow for successful randconfig builds that use COMPILE_TEST. Reported-by: Jim Davis <jim.epost(a)gmail.com> Fixes: dc1ebd1811e9 ("ASoC: qcom: Add apq8016 lpass driver support") Fixes: 9bae4880acee ("ASoC: qcom: move ipq806x specific bits out of lpass driver") Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- Created fix in response to Jim's bug report: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-May/092870.html sound/soc/qcom/Kconfig | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 938144c..290b056 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -14,13 +14,11 @@ config SND_SOC_LPASS_PLATFORM config SND_SOC_LPASS_IPQ806X tristate - depends on SND_SOC_QCOM select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM config SND_SOC_LPASS_APQ8016 tristate - depends on SND_SOC_QCOM select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [asoc:topic/arizona 1/7] sound/soc/codecs/arizona.h:122:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS'
by kbuild test robot 03 Jun '15

03 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/arizona head: 69a6582eeb17dc083b2510f1ca2eaa54ff679b49 commit: 21a37e39e02d7f57691219fee88cf1d48a74e5bd [1/7] ASoC: arizona: Use auto disable muxes for routing config: xtensa-allyesconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 21a37e39e02d7f57691219fee88cf1d48a74e5bd # save the attached .config to linux build tree make.cross ARCH=xtensa All error/warnings: In file included from sound/soc/codecs/wm5102.c:33:0: >> include/linux/mfd/arizona/registers.h:646:50: error: expected ')' before numeric constant #define ARIZONA_EQ1MIX_INPUT_1_SOURCE 0x880 ^ sound/soc/codecs/arizona.h:111:9: note: in definition of macro 'ARIZONA_MUX_ENUM_DECL' name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values) ^ >> sound/soc/codecs/arizona.h:122:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in1, base_reg); \ ^ >> sound/soc/codecs/wm5102.c:943:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ >> sound/soc/codecs/wm5102.c:943:26: note: in expansion of macro 'ARIZONA_EQ1MIX_INPUT_1_SOURCE' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from include/sound/soc.h:365:0, from sound/soc/codecs/wm5102.c:24: >> sound/soc/codecs/wm5102.c:943:21: error: 'EQ1_in1_enum' undeclared here (not in a function) ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ include/sound/soc-dapm.h:298:37: note: in definition of macro 'SOC_DAPM_ENUM' .private_value = (unsigned long)&xenum } ^ >> sound/soc/codecs/arizona.h:119:9: note: in expansion of macro 'ARIZONA_MUX_CTL_DECL' static ARIZONA_MUX_CTL_DECL(name) ^ >> sound/soc/codecs/arizona.h:122:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in1, base_reg); \ ^ >> sound/soc/codecs/wm5102.c:943:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from sound/soc/codecs/wm5102.c:33:0: >> include/linux/mfd/arizona/registers.h:646:50: error: expected ')' before numeric constant #define ARIZONA_EQ1MIX_INPUT_1_SOURCE 0x880 ^ sound/soc/codecs/arizona.h:111:9: note: in definition of macro 'ARIZONA_MUX_ENUM_DECL' name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values) ^ sound/soc/codecs/arizona.h:123:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in2, base_reg + 2); \ ^ >> sound/soc/codecs/wm5102.c:943:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ >> sound/soc/codecs/wm5102.c:943:26: note: in expansion of macro 'ARIZONA_EQ1MIX_INPUT_1_SOURCE' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from include/sound/soc.h:365:0, from sound/soc/codecs/wm5102.c:24: >> sound/soc/codecs/wm5102.c:943:21: error: 'EQ1_in2_enum' undeclared here (not in a function) ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ include/sound/soc-dapm.h:298:37: note: in definition of macro 'SOC_DAPM_ENUM' .private_value = (unsigned long)&xenum } ^ >> sound/soc/codecs/arizona.h:119:9: note: in expansion of macro 'ARIZONA_MUX_CTL_DECL' static ARIZONA_MUX_CTL_DECL(name) ^ sound/soc/codecs/arizona.h:123:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in2, base_reg + 2); \ ^ >> sound/soc/codecs/wm5102.c:943:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from sound/soc/codecs/wm5102.c:33:0: >> include/linux/mfd/arizona/registers.h:646:50: error: expected ')' before numeric constant #define ARIZONA_EQ1MIX_INPUT_1_SOURCE 0x880 ^ sound/soc/codecs/arizona.h:111:9: note: in definition of macro 'ARIZONA_MUX_ENUM_DECL' name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values) ^ sound/soc/codecs/arizona.h:124:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in3, base_reg + 4); \ ^ >> sound/soc/codecs/wm5102.c:943:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ >> sound/soc/codecs/wm5102.c:943:26: note: in expansion of macro 'ARIZONA_EQ1MIX_INPUT_1_SOURCE' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from include/sound/soc.h:365:0, from sound/soc/codecs/wm5102.c:24: >> sound/soc/codecs/wm5102.c:943:21: error: 'EQ1_in3_enum' undeclared here (not in a function) ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ include/sound/soc-dapm.h:298:37: note: in definition of macro 'SOC_DAPM_ENUM' .private_value = (unsigned long)&xenum } ^ >> sound/soc/codecs/arizona.h:119:9: note: in expansion of macro 'ARIZONA_MUX_CTL_DECL' static ARIZONA_MUX_CTL_DECL(name) ^ sound/soc/codecs/arizona.h:124:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in3, base_reg + 4); \ ^ >> sound/soc/codecs/wm5102.c:943:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from sound/soc/codecs/wm5102.c:33:0: -- In file included from sound/soc/codecs/wm5110.c:32:0: >> include/linux/mfd/arizona/registers.h:646:50: error: expected ')' before numeric constant #define ARIZONA_EQ1MIX_INPUT_1_SOURCE 0x880 ^ sound/soc/codecs/arizona.h:111:9: note: in definition of macro 'ARIZONA_MUX_ENUM_DECL' name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values) ^ >> sound/soc/codecs/arizona.h:122:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in1, base_reg); \ ^ >> sound/soc/codecs/wm5110.c:470:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ >> sound/soc/codecs/wm5110.c:470:26: note: in expansion of macro 'ARIZONA_EQ1MIX_INPUT_1_SOURCE' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from include/sound/soc.h:365:0, from sound/soc/codecs/wm5110.c:24: >> sound/soc/codecs/wm5110.c:470:21: error: 'EQ1_in1_enum' undeclared here (not in a function) ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ include/sound/soc-dapm.h:298:37: note: in definition of macro 'SOC_DAPM_ENUM' .private_value = (unsigned long)&xenum } ^ >> sound/soc/codecs/arizona.h:119:9: note: in expansion of macro 'ARIZONA_MUX_CTL_DECL' static ARIZONA_MUX_CTL_DECL(name) ^ >> sound/soc/codecs/arizona.h:122:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in1, base_reg); \ ^ >> sound/soc/codecs/wm5110.c:470:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from sound/soc/codecs/wm5110.c:32:0: >> include/linux/mfd/arizona/registers.h:646:50: error: expected ')' before numeric constant #define ARIZONA_EQ1MIX_INPUT_1_SOURCE 0x880 ^ sound/soc/codecs/arizona.h:111:9: note: in definition of macro 'ARIZONA_MUX_ENUM_DECL' name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values) ^ sound/soc/codecs/arizona.h:123:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in2, base_reg + 2); \ ^ >> sound/soc/codecs/wm5110.c:470:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ >> sound/soc/codecs/wm5110.c:470:26: note: in expansion of macro 'ARIZONA_EQ1MIX_INPUT_1_SOURCE' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from include/sound/soc.h:365:0, from sound/soc/codecs/wm5110.c:24: >> sound/soc/codecs/wm5110.c:470:21: error: 'EQ1_in2_enum' undeclared here (not in a function) ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ include/sound/soc-dapm.h:298:37: note: in definition of macro 'SOC_DAPM_ENUM' .private_value = (unsigned long)&xenum } ^ >> sound/soc/codecs/arizona.h:119:9: note: in expansion of macro 'ARIZONA_MUX_CTL_DECL' static ARIZONA_MUX_CTL_DECL(name) ^ sound/soc/codecs/arizona.h:123:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in2, base_reg + 2); \ ^ >> sound/soc/codecs/wm5110.c:470:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from sound/soc/codecs/wm5110.c:32:0: >> include/linux/mfd/arizona/registers.h:646:50: error: expected ')' before numeric constant #define ARIZONA_EQ1MIX_INPUT_1_SOURCE 0x880 ^ sound/soc/codecs/arizona.h:111:9: note: in definition of macro 'ARIZONA_MUX_ENUM_DECL' name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values) ^ sound/soc/codecs/arizona.h:124:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in3, base_reg + 4); \ ^ >> sound/soc/codecs/wm5110.c:470:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ >> sound/soc/codecs/wm5110.c:470:26: note: in expansion of macro 'ARIZONA_EQ1MIX_INPUT_1_SOURCE' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from include/sound/soc.h:365:0, from sound/soc/codecs/wm5110.c:24: >> sound/soc/codecs/wm5110.c:470:21: error: 'EQ1_in3_enum' undeclared here (not in a function) ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ include/sound/soc-dapm.h:298:37: note: in definition of macro 'SOC_DAPM_ENUM' .private_value = (unsigned long)&xenum } ^ >> sound/soc/codecs/arizona.h:119:9: note: in expansion of macro 'ARIZONA_MUX_CTL_DECL' static ARIZONA_MUX_CTL_DECL(name) ^ sound/soc/codecs/arizona.h:124:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in3, base_reg + 4); \ ^ >> sound/soc/codecs/wm5110.c:470:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from sound/soc/codecs/wm5110.c:32:0: -- In file included from sound/soc/codecs/wm8997.c:32:0: >> include/linux/mfd/arizona/registers.h:646:50: error: expected ')' before numeric constant #define ARIZONA_EQ1MIX_INPUT_1_SOURCE 0x880 ^ sound/soc/codecs/arizona.h:111:9: note: in definition of macro 'ARIZONA_MUX_ENUM_DECL' name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values) ^ >> sound/soc/codecs/arizona.h:122:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in1, base_reg); \ ^ >> sound/soc/codecs/wm8997.c:333:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ >> sound/soc/codecs/wm8997.c:333:26: note: in expansion of macro 'ARIZONA_EQ1MIX_INPUT_1_SOURCE' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from include/sound/soc.h:365:0, from sound/soc/codecs/wm8997.c:24: >> sound/soc/codecs/wm8997.c:333:21: error: 'EQ1_in1_enum' undeclared here (not in a function) ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ include/sound/soc-dapm.h:298:37: note: in definition of macro 'SOC_DAPM_ENUM' .private_value = (unsigned long)&xenum } ^ >> sound/soc/codecs/arizona.h:119:9: note: in expansion of macro 'ARIZONA_MUX_CTL_DECL' static ARIZONA_MUX_CTL_DECL(name) ^ >> sound/soc/codecs/arizona.h:122:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in1, base_reg); \ ^ >> sound/soc/codecs/wm8997.c:333:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from sound/soc/codecs/wm8997.c:32:0: >> include/linux/mfd/arizona/registers.h:646:50: error: expected ')' before numeric constant #define ARIZONA_EQ1MIX_INPUT_1_SOURCE 0x880 ^ sound/soc/codecs/arizona.h:111:9: note: in definition of macro 'ARIZONA_MUX_ENUM_DECL' name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values) ^ sound/soc/codecs/arizona.h:123:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in2, base_reg + 2); \ ^ >> sound/soc/codecs/wm8997.c:333:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ >> sound/soc/codecs/wm8997.c:333:26: note: in expansion of macro 'ARIZONA_EQ1MIX_INPUT_1_SOURCE' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from include/sound/soc.h:365:0, from sound/soc/codecs/wm8997.c:24: >> sound/soc/codecs/wm8997.c:333:21: error: 'EQ1_in2_enum' undeclared here (not in a function) ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ include/sound/soc-dapm.h:298:37: note: in definition of macro 'SOC_DAPM_ENUM' .private_value = (unsigned long)&xenum } ^ >> sound/soc/codecs/arizona.h:119:9: note: in expansion of macro 'ARIZONA_MUX_CTL_DECL' static ARIZONA_MUX_CTL_DECL(name) ^ sound/soc/codecs/arizona.h:123:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in2, base_reg + 2); \ ^ >> sound/soc/codecs/wm8997.c:333:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from sound/soc/codecs/wm8997.c:32:0: >> include/linux/mfd/arizona/registers.h:646:50: error: expected ')' before numeric constant #define ARIZONA_EQ1MIX_INPUT_1_SOURCE 0x880 ^ sound/soc/codecs/arizona.h:111:9: note: in definition of macro 'ARIZONA_MUX_ENUM_DECL' name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values) ^ sound/soc/codecs/arizona.h:124:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in3, base_reg + 4); \ ^ >> sound/soc/codecs/wm8997.c:333:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ >> sound/soc/codecs/wm8997.c:333:26: note: in expansion of macro 'ARIZONA_EQ1MIX_INPUT_1_SOURCE' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from include/sound/soc.h:365:0, from sound/soc/codecs/wm8997.c:24: >> sound/soc/codecs/wm8997.c:333:21: error: 'EQ1_in3_enum' undeclared here (not in a function) ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ include/sound/soc-dapm.h:298:37: note: in definition of macro 'SOC_DAPM_ENUM' .private_value = (unsigned long)&xenum } ^ >> sound/soc/codecs/arizona.h:119:9: note: in expansion of macro 'ARIZONA_MUX_CTL_DECL' static ARIZONA_MUX_CTL_DECL(name) ^ sound/soc/codecs/arizona.h:124:2: note: in expansion of macro 'ARIZONA_MUX_ENUMS' ARIZONA_MUX_ENUMS(name##_in3, base_reg + 4); \ ^ >> sound/soc/codecs/wm8997.c:333:1: note: in expansion of macro 'ARIZONA_MIXER_ENUMS' ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ^ In file included from sound/soc/codecs/wm8997.c:32:0: vim +/ARIZONA_MUX_ENUMS +122 sound/soc/codecs/arizona.h 07ed873e Mark Brown 2012-06-18 105 SOC_SINGLE_RANGE_TLV(name " Input 4 Volume", base + 7, \ 07ed873e Mark Brown 2012-06-18 106 ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ 07ed873e Mark Brown 2012-06-18 107 arizona_mixer_tlv) 07ed873e Mark Brown 2012-06-18 108 07ed873e Mark Brown 2012-06-18 109 #define ARIZONA_MUX_ENUM_DECL(name, reg) \ 21a37e39 Charles Keepax 2015-05-01 110 SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL( \ 21a37e39 Charles Keepax 2015-05-01 @111 name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values) 07ed873e Mark Brown 2012-06-18 112 07ed873e Mark Brown 2012-06-18 113 #define ARIZONA_MUX_CTL_DECL(name) \ 07ed873e Mark Brown 2012-06-18 114 const struct snd_kcontrol_new name##_mux = \ 355e3a08 Lars-Peter Clausen 2014-04-14 115 SOC_DAPM_ENUM("Route", name##_enum) 07ed873e Mark Brown 2012-06-18 116 17bd09e5 Charles Keepax 2012-11-13 117 #define ARIZONA_MUX_ENUMS(name, base_reg) \ 17bd09e5 Charles Keepax 2012-11-13 118 static ARIZONA_MUX_ENUM_DECL(name##_enum, base_reg); \ 17bd09e5 Charles Keepax 2012-11-13 @119 static ARIZONA_MUX_CTL_DECL(name) 17bd09e5 Charles Keepax 2012-11-13 120 07ed873e Mark Brown 2012-06-18 121 #define ARIZONA_MIXER_ENUMS(name, base_reg) \ 17bd09e5 Charles Keepax 2012-11-13 @122 ARIZONA_MUX_ENUMS(name##_in1, base_reg); \ 17bd09e5 Charles Keepax 2012-11-13 123 ARIZONA_MUX_ENUMS(name##_in2, base_reg + 2); \ 17bd09e5 Charles Keepax 2012-11-13 124 ARIZONA_MUX_ENUMS(name##_in3, base_reg + 4); \ 17bd09e5 Charles Keepax 2012-11-13 125 ARIZONA_MUX_ENUMS(name##_in4, base_reg + 6) :::::: The code at line 122 was first introduced by commit :::::: 17bd09e5457740425fee6ca2bb685a2a55c12c18 ASoC: arizona: Add support for multiplexer with no associated mixer :::::: TO: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> :::::: CC: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

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[alsa-devel] [PATCH v4 0/3] ASoC: qcom: add support to apq8016 audio
by Srinivas Kandagatla 03 Jun '15

03 Jun '15

Hi Mark, Thanks for applying the patches. Am resending the leftover patches rebased on top of topic/qcom branch. I have funtionally tested these patches on AP8016-SBC board and also compile tested them without CONFIG_OF on x86 to make sure nothing breaks. All these patches are tested and Acked. I Also included Arnd's Patch to fix Kconfig dependency on STORM board. APQ8016 has 4 MI2S( Primary, Secondary, Tertiary, Quaternary) which can be routed to internal wcd codec or external codecs. This routing is controlled by 2 mux registers. Changes since v3 - dropped OF dependency patch - dropped 6 patches which are already applied by Mark. - fixed Kconfig as suggested by Arnd. Changes since v2(https://lkml.org/lkml/2015/5/16/236) - dropped 3 patches which are already applied by Mark. - rebased on top of topic/qcom branch. - changed dev_err to dev_warn spotted by Kenneth. - fixed typo spotted by kenneth. Changes since v1(https://lwn.net/Articles/644266/) - Fixed Kconfig dependencies spotted by Kenneth - Fixed compilation error on ipq806x spotted by Kenneth - Fixed clk error messages spotted by Kenneth Changes since RFC(https://lwn.net/Articles/642661/) - droped cleanup patches, as they are already applied by Mark. - prefixed shared functions with asoc_qcom_* as suggested by Kenneth - made LPASS driver depended on OF. - removed ipq806x specific bits from ipaif_reg.h - removed no_osr clk flag as suggested by Mark and Kenneth - fixed irq return values spotted by Kenneth - renamed the machine file to apq8016-sbc.c as suggested by Kenneth. - use dai->driver->id instead of dai->id as it breaks on Storm board. --srini Arnd Bergmann (1): ASoC: qcom: fix STORM board Kconfig Srinivas Kandagatla (2): ASoC: qcom: Add apq8016 lpass driver support ASoC: qcom: add apq8016 sound card support include/dt-bindings/sound/apq8016-lpass.h | 9 ++ sound/soc/qcom/Kconfig | 17 ++- sound/soc/qcom/Makefile | 4 + sound/soc/qcom/apq8016_sbc.c | 215 ++++++++++++++++++++++++++ sound/soc/qcom/lpass-apq8016.c | 242 ++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 + 6 files changed, 490 insertions(+), 1 deletion(-) create mode 100644 include/dt-bindings/sound/apq8016-lpass.h create mode 100644 sound/soc/qcom/apq8016_sbc.c create mode 100644 sound/soc/qcom/lpass-apq8016.c -- 1.9.1

3 9

[alsa-devel] [ALSA-UTILS][PATCH] Add support for cplay and crecord
by Qais Yousef 03 Jun '15

03 Jun '15

cplay and crecord use compress offload API to play and record compressed audio. They're based on cplay and crec from tinycompress library using LGPL license. For now cplay only supports playing mp3 files. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- I renamed crec to crecord also to match aplay and arecord, hopefully you don't mind Vinod. This patch is dependent on my other patch that adds support for compress offload to alsa-lib. I needed to include <sound/compress_params.h> in cplay.c and crec.c but I couldn't find an example of any C file which directly includes <sound/*.h> The norm seems to be to just include <alsa/asoundlib.h>. Do I need to redefine structs from <sound/compress_params.h> to newly added <alsa/compress.h>? <alsa/pcm.h> seems to redefine structs from <sound/asound.h>. I could only test cplay but have no means to test crecord at the moment. Makefile.am | 3 + configure.ac | 6 +- cplay/Makefile.am | 14 ++ cplay/cplay.c | 294 +++++++++++++++++++++++++++++++++++ cplay/crec.c | 449 ++++++++++++++++++++++++++++++++++++++++++++++++++++++ cplay/tinymp3.h | 72 +++++++++ 6 files changed, 837 insertions(+), 1 deletion(-) create mode 100644 cplay/Makefile.am create mode 100644 cplay/cplay.c create mode 100644 cplay/crec.c create mode 100644 cplay/tinymp3.h diff --git a/Makefile.am b/Makefile.am index 5bbe588a8d84..0842657530fd 100644 --- a/Makefile.am +++ b/Makefile.am @@ -19,6 +19,9 @@ if ALSALOOP SUBDIRS += alsaloop endif endif +if HAVE_COMPRESS +SUBDIRS += cplay +endif if HAVE_SEQ SUBDIRS += seq endif diff --git a/configure.ac b/configure.ac index f09aa5484d1d..c08c24b90658 100644 --- a/configure.ac +++ b/configure.ac @@ -42,6 +42,8 @@ fi dnl Check components AC_CHECK_HEADERS([alsa/pcm.h], [have_pcm="yes"], [have_pcm="no"], [#include <alsa/asoundlib.h>]) +AC_CHECK_HEADERS([alsa/compress.h], [have_compress="yes"], [have_compress="no"], + [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([alsa/mixer.h], [have_mixer="yes"], [have_mixer="no"], [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([alsa/rawmidi.h], [have_rawmidi="yes"], [have_rawmidi="no"], @@ -54,6 +56,7 @@ AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"] [#include <samplerate.h>]) AM_CONDITIONAL(HAVE_PCM, test "$have_pcm" = "yes") +AM_CONDITIONAL(HAVE_COMPRESS, test "$have_compress" = "yes") AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") AM_CONDITIONAL(HAVE_SEQ, test "$have_seq" = "yes") @@ -359,7 +362,8 @@ AC_OUTPUT(Makefile alsactl/Makefile alsactl/init/Makefile \ alsaconf/alsaconf alsaconf/Makefile \ alsaconf/po/Makefile \ alsaucm/Makefile \ - aplay/Makefile include/Makefile iecset/Makefile utils/Makefile \ + aplay/Makefile cplay/Makefile \ + include/Makefile iecset/Makefile utils/Makefile \ utils/alsa-utils.spec seq/Makefile seq/aconnect/Makefile \ seq/aplaymidi/Makefile seq/aseqdump/Makefile seq/aseqnet/Makefile \ speaker-test/Makefile speaker-test/samples/Makefile \ diff --git a/cplay/Makefile.am b/cplay/Makefile.am new file mode 100644 index 000000000000..bcb1bfce7a2d --- /dev/null +++ b/cplay/Makefile.am @@ -0,0 +1,14 @@ +LIBRT = @LIBRT@ + +AM_CPPFLAGS = -I$(top_srcdir)/include +LDADD = $(LIBINTL) $(LIBRT) + +# debug flags +#LDFLAGS = -static +#LDADD += -ldl + +bin_PROGRAMS = cplay crecord +noinst_HEADERS = tinymp3.h + +cplay_SOURCES = cplay.c +crecord_SOURCES = crec.c diff --git a/cplay/cplay.c b/cplay/cplay.c new file mode 100644 index 000000000000..a017394e5b4f --- /dev/null +++ b/cplay/cplay.c @@ -0,0 +1,294 @@ +/* + * tinyplay command line player for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#include <alsa/asoundlib.h> +#include <sound/compress_params.h> + +#include "tinymp3.h" + +static int verbose; + +static void usage(void) +{ + fprintf(stderr, "usage: cplay [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n\n" + "-v\tverbose mode\n" + "-h\tPrints this help list\n\n" + "Example:\n" + "\tcplay -c 1 -d 2 test.mp3\n" + "\tcplay -f 5 test.mp3\n"); + + exit(EXIT_FAILURE); +} + +void play_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag); + +struct mp3_header { + uint16_t sync; + uint8_t format1; + uint8_t format2; +}; + +int parse_mp3_header(struct mp3_header *header, unsigned int *num_channels, + unsigned int *sample_rate, unsigned int *bit_rate) +{ + int ver_idx, mp3_version, layer, bit_rate_idx, sample_rate_idx, channel_idx; + + /* check sync bits */ + if ((header->sync & MP3_SYNC) != MP3_SYNC) { + fprintf(stderr, "Error: Can't find sync word\n"); + return -1; + } + ver_idx = (header->sync >> 11) & 0x03; + mp3_version = ver_idx == 0 ? MPEG25 : ((ver_idx & 0x1) ? MPEG1 : MPEG2); + layer = 4 - ((header->sync >> 9) & 0x03); + bit_rate_idx = ((header->format1 >> 4) & 0x0f); + sample_rate_idx = ((header->format1 >> 2) & 0x03); + channel_idx = ((header->format2 >> 6) & 0x03); + + if (sample_rate_idx == 3 || layer == 4 || bit_rate_idx == 15) { + fprintf(stderr, "Error: Can't find valid header\n"); + return -1; + } + *num_channels = (channel_idx == MONO ? 1 : 2); + *sample_rate = mp3_sample_rates[mp3_version][sample_rate_idx]; + *bit_rate = (mp3_bit_rates[mp3_version][layer - 1][bit_rate_idx]) * 1000; + if (verbose) + printf("%s: exit\n", __func__); + return 0; +} + +int check_codec_format_supported(unsigned int card, unsigned int device, struct snd_codec *codec) +{ + if (snd_compr_is_codec_supported(card, device, COMPRESS_IN, codec) == false) { + fprintf(stderr, "Error: This codec or format is not supported by DSP\n"); + return -1; + } + return 0; +} + +static int print_time(struct snd_compr *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (snd_compr_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + return -1; + } else + fprintf(stderr, "DSP played %jd.%jd\n", (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + return 0; +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0; + + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "hvb:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + play_samples(file, card, device, buffer_size, frag); + + fprintf(stderr, "Finish Playing.... Close Normally\n"); + exit(EXIT_SUCCESS); +} + +void play_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag) +{ + struct snd_compr_config config; + struct snd_codec codec; + struct snd_compr *compress; + struct mp3_header header; + FILE *file; + char *buffer; + int size, num_read, wrote; + unsigned int channels, rate, bits; + + if (verbose) + printf("%s: entry\n", __func__); + file = fopen(name, "rb"); + if (!file) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + fread(&header, sizeof(header), 1, file); + + if (parse_mp3_header(&header, &channels, &rate, &bits) == -1) { + fclose(file); + exit(EXIT_FAILURE); + } + + codec.id = SND_AUDIOCODEC_MP3; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = rate; + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + fclose(file); + exit(EXIT_FAILURE); + } + codec.bit_rate = bits; + codec.rate_control = 0; + codec.profile = 0; + codec.level = 0; + codec.ch_mode = 0; + codec.format = 0; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } else { + /* use driver defaults */ + config.fragment_size = 0; + config.fragments = 0; + } + config.codec = &codec; + + compress = snd_compr_open(card, device, COMPRESS_IN, &config); + if (!compress || !snd_compr_is_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + goto FILE_EXIT; + }; + if (verbose) + printf("%s: Opened compress device\n", __func__); + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto COMP_EXIT; + } + + /* we will write frag fragment_size and then start */ + num_read = fread(buffer, 1, size * config.fragments, file); + if (num_read > 0) { + if (verbose) + printf("%s: Doing first buffer write of %d\n", __func__, num_read); + wrote = snd_compr_write(compress, buffer, num_read); + if (wrote < 0) { + fprintf(stderr, "Error %d playing sample\n", wrote); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + goto BUF_EXIT; + } + if (wrote != num_read) { + /* TODO: Buufer pointer needs to be set here */ + fprintf(stderr, "We wrote %d, DSP accepted %d\n", num_read, wrote); + } + } + printf("Playing file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz, Bit Rate %d\n", + SND_AUDIOCODEC_MP3, channels, rate, bits); + + snd_compr_start(compress); + if (verbose) + printf("%s: You should hear audio NOW!!!\n", __func__); + + do { + num_read = fread(buffer, 1, size, file); + if (num_read > 0) { + wrote = snd_compr_write(compress, buffer, num_read); + if (wrote < 0) { + fprintf(stderr, "Error playing sample\n"); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + goto BUF_EXIT; + } + if (wrote != num_read) { + /* TODO: Buffer pointer needs to be set here */ + fprintf(stderr, "We wrote %d, DSP accepted %d\n", num_read, wrote); + } + if (verbose) { + print_time(compress); + printf("%s: wrote %d\n", __func__, wrote); + } + } + } while (num_read > 0); + + if (verbose) + printf("%s: exit success\n", __func__); + /* issue drain if it supports */ + snd_compr_drain(compress); + free(buffer); + fclose(file); + snd_compr_close(compress); + return; +BUF_EXIT: + free(buffer); +COMP_EXIT: + snd_compr_close(compress); +FILE_EXIT: + fclose(file); + if (verbose) + printf("%s: exit failure\n", __func__); + exit(EXIT_FAILURE); +} + diff --git a/cplay/crec.c b/cplay/crec.c new file mode 100644 index 000000000000..6a3b4a260ceb --- /dev/null +++ b/cplay/crec.c @@ -0,0 +1,449 @@ +/* + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013-2014, Wolfson Microelectronic Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <alsa/asoundlib.h> +#include <sound/compress_params.h> + +static int verbose; +static int file; +static FILE *finfo; +static bool streamed; + +static const unsigned int DEFAULT_CHANNELS = 1; +static const unsigned int DEFAULT_RATE = 44100; +static const unsigned int DEFAULT_FORMAT = SND_PCM_FORMAT_S16_LE; + +struct riff_chunk { + char desc[4]; + uint32_t size; +} __attribute__((__packed__)); + +struct wave_header { + struct { + struct riff_chunk chunk; + char format[4]; + } __attribute__((__packed__)) riff; + + struct { + struct riff_chunk chunk; + uint16_t type; + uint16_t channels; + uint32_t rate; + uint32_t byterate; + uint16_t blockalign; + uint16_t samplebits; + } __attribute__((__packed__)) fmt; + + struct { + struct riff_chunk chunk; + } __attribute__((__packed__)) data; +} __attribute__((__packed__)); + +const struct wave_header blank_wave_header = { + .riff = { + .chunk = { + .desc = "RIFF", + }, + .format = "WAVE", + }, + .fmt = { + .chunk = { + .desc = "fmt ", /* Note the space is important here */ + .size = sizeof(blank_wave_header.fmt) - + sizeof(blank_wave_header.fmt.chunk), + }, + .type = 0x01, /* PCM */ + }, + .data = { + .chunk = { + .desc = "data", + }, + }, +}; + +static void init_wave_header(struct wave_header *header, uint16_t channels, + uint32_t rate, uint16_t samplebits) +{ + memcpy(header, &blank_wave_header, sizeof(blank_wave_header)); + + header->fmt.channels = channels; + header->fmt.rate = rate; + header->fmt.byterate = channels * rate * (samplebits / 8); + header->fmt.blockalign = channels * (samplebits / 8); + header->fmt.samplebits = samplebits; +} + +static void size_wave_header(struct wave_header *header, uint32_t size) +{ + header->riff.chunk.size = sizeof(*header) - + sizeof(header->riff.chunk) + size; + header->data.chunk.size = size; +} + +static void usage(void) +{ + fprintf(stderr, "usage: crecord [OPTIONS] [filename]\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n" + "-v\tverbose mode\n" + "-l\tlength of record in seconds\n" + "-h\tPrints this help list\n\n" + "-C\tSpecify the number of channels (default %u)\n" + "-R\tSpecify the sample rate (default %u)\n" + "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "If filename is not given the output is\n" + "written to stdout\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.wav\n" + "\tcrec -f 5 test.wav\n", + DEFAULT_CHANNELS, DEFAULT_RATE); + + exit(EXIT_FAILURE); +} + +static int print_time(struct snd_compr *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (snd_compr_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + return -1; + } else { + fprintf(finfo, "DSP recorded %jd.%jd\n", + (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + } + return 0; +} + +static int finish_record() +{ + struct wave_header header; + int ret; + size_t nread, written; + + if (!file) + return -ENOENT; + + /* can't rewind if streaming to stdout */ + if (streamed) + return 0; + + /* Get amount of data written to file */ + ret = lseek(file, 0, SEEK_END); + if (ret < 0) + return -errno; + + written = ret; + if (written < sizeof(header)) + return -ENOENT; + written -= sizeof(header); + + /* Sync file header from file */ + ret = lseek(file, 0, SEEK_SET); + if (ret < 0) + return -errno; + + nread = read(file, &header, sizeof(header)); + if (nread != sizeof(header)) + return -errno; + + /* Update file header */ + ret = lseek(file, 0, SEEK_SET); + if (ret < 0) + return -errno; + + size_wave_header(&header, written); + + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) + return -errno; + + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, + unsigned int length, unsigned int rate, + unsigned int channels, unsigned int format) +{ + struct snd_compr_config config; + struct snd_codec codec; + struct snd_compr *compress; + struct wave_header header; + char *buffer; + size_t written; + int read, ret; + unsigned int size, total_read = 0; + unsigned int samplebits; + + switch (format) { + case SND_PCM_FORMAT_S32_LE: + samplebits = 32; + break; + default: + samplebits = 16; + break; + } + + /* Convert length from seconds to bytes */ + length = length * rate * (samplebits / 8) * channels; + + if (verbose) + fprintf(finfo, "%s: entry, reading %u bytes\n", __func__, length); + if (!name) { + file = STDOUT_FILENO; + } else { + file = open(name, O_RDWR | O_CREAT, S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP); + if (file == -1) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + } + + /* Write a header, will update with size once record is complete */ + if (!streamed) { + init_wave_header(&header, channels, rate, samplebits); + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) { + fprintf(stderr, "Error writing output file header: %s\n", + strerror(errno)); + goto file_exit; + } + } + + memset(&codec, 0, sizeof(codec)); + memset(&config, 0, sizeof(config)); + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = rate; + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + goto file_exit; + } + codec.format = format; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } + config.codec = &codec; + + compress = snd_compr_open(card, device, COMPRESS_OUT, &config); + if (!compress || !snd_compr_is_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + goto file_exit; + }; + + if (verbose) + fprintf(finfo, "%s: Opened compress device\n", __func__); + + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto comp_exit; + } + + fprintf(finfo, "Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + fprintf(finfo, "Codec %u Format %u Channels %u, %u Hz\n", + codec.id, codec.format, codec.ch_out, rate); + + snd_compr_start(compress); + + if (verbose) + fprintf(finfo, "%s: Capturing audio NOW!!!\n", __func__); + + do { + if (length && size > length - total_read) + size = length - total_read; + + read = snd_compr_read(compress, buffer, size); + if (read < 0) { + fprintf(stderr, "Error reading sample\n"); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + goto buf_exit; + } + if ((unsigned int)read != size) { + fprintf(stderr, "We read %d, DSP sent %d\n", + size, read); + } + + if (read > 0) { + total_read += read; + + written = write(file, buffer, read); + if (written != (size_t)read) { + fprintf(stderr, "Error writing output file: %s\n", + strerror(errno)); + goto buf_exit; + } + if (verbose) { + print_time(compress); + fprintf(finfo, "%s: read %d\n", __func__, read); + } + } + } while (!length || total_read < length); + + ret = snd_compr_stop(compress); + if (ret < 0) { + fprintf(stderr, "Error closing stream\n"); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + } + + ret = finish_record(); + if (ret < 0) { + fprintf(stderr, "Failed to finish header: %s\n", strerror(ret)); + goto buf_exit; + } + + if (verbose) + fprintf(finfo, "%s: exit success\n", __func__); + + free(buffer); + close(file); + file = 0; + + snd_compr_close(compress); + + return; +buf_exit: + free(buffer); +comp_exit: + snd_compr_close(compress); +file_exit: + close(file); + + if (verbose) + fprintf(finfo, "%s: exit failure\n", __func__); + + exit(EXIT_FAILURE); +} + +static void sig_handler(int signum __attribute__ ((unused))) +{ + finish_record(); + + if (file) + close(file); + + _exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, length = 0; + unsigned int rate = DEFAULT_RATE, channels = DEFAULT_CHANNELS; + unsigned int format = DEFAULT_FORMAT; + + if (signal(SIGINT, sig_handler) == SIG_ERR) { + fprintf(stderr, "Error registering signal handler\n"); + exit(EXIT_FAILURE); + } + + if (argc < 1) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "hvl:R:C:F:b:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + case 'l': + length = strtol(optarg, NULL, 10); + break; + case 'R': + rate = strtol(optarg, NULL, 10); + break; + case 'C': + channels = strtol(optarg, NULL, 10); + break; + case 'F': + if (strcmp(optarg, "S16_LE") == 0) { + format = SND_PCM_FORMAT_S16_LE; + } else if (strcmp(optarg, "S32_LE") == 0) { + format = SND_PCM_FORMAT_S32_LE; + } else { + fprintf(stderr, "Unrecognised format: %s\n", + optarg); + usage(); + } + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) { + file = NULL; + finfo = fopen("/dev/null", "w"); + streamed = true; + } else { + file = argv[optind]; + finfo = stdout; + streamed = false; + } + + capture_samples(file, card, device, buffer_size, frag, length, + rate, channels, format); + + fprintf(finfo, "Finish capturing... Close Normally\n"); + + exit(EXIT_SUCCESS); +} + diff --git a/cplay/tinymp3.h b/cplay/tinymp3.h new file mode 100644 index 000000000000..13afaec2757d --- /dev/null +++ b/cplay/tinymp3.h @@ -0,0 +1,72 @@ +/* + * mp3 header and parsing + * Copyright (c) 2011-2012, Intel Corporation. + * + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + + +#ifndef __TINYMP3_H +#define __TINYMP3_H + +#if defined(__cplusplus) +extern "C" { +#endif + + +#define MP3_SYNC 0xe0ff + +const int mp3_sample_rates[3][3] = { + {44100, 48000, 32000}, /* MPEG-1 */ + {22050, 24000, 16000}, /* MPEG-2 */ + {11025, 12000, 8000}, /* MPEG-2.5 */ +}; + +const int mp3_bit_rates[3][3][15] = { + { + /* MPEG-1 */ + { 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448}, /* Layer 1 */ + { 0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384}, /* Layer 2 */ + { 0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320}, /* Layer 3 */ + }, + { + /* MPEG-2 */ + { 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256}, /* Layer 1 */ + { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}, /* Layer 2 */ + { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}, /* Layer 3 */ + }, + { + /* MPEG-2.5 */ + { 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256}, /* Layer 1 */ + { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}, /* Layer 2 */ + { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}, /* Layer 3 */ + }, +}; + +enum mpeg_version { + MPEG1 = 0, + MPEG2 = 1, + MPEG25 = 2 +}; + +enum mp3_stereo_mode { + STEREO = 0x00, + JOINT = 0x01, + DUAL = 0x02, + MONO = 0x03 +}; + +#endif -- 2.1.0

5 19

Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS
by Clemens Ladisch 03 Jun '15

03 Jun '15

Srinivasan S wrote: > $ aplay -f dat -D VOUTL new.wav > Playing WAVE 'new.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo > aplay: set_params:1087: Channels count non available You are trying to play a two-channel file on a single-channel device. Regards, Clemens

2 16

[alsa-devel] [PATCH] ALSA: hda - remove controller dependency on i915 power well for SKL
by libin.yang＠intel.com 03 Jun '15

03 Jun '15

From: Libin Yang <libin.yang(a)intel.com> For SKL, only the HDMI codec is in the display power well while the HD-A controller isn't. So the controller flag 'need_i915_power' is not set to release the display power after probe, and the codec flag 'link_power_control" is set to request/release the display power via bus link_power ops. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/hda_intel.c | 9 +++++++-- sound/pci/hda/patch_hdmi.c | 2 +- 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 391e4f8..e5597fd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -345,6 +345,11 @@ enum { #define use_vga_switcheroo(chip) 0 #endif +#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ + ((pci)->device == 0x0c0c) || \ + ((pci)->device == 0x0d0c) || \ + ((pci)->device == 0x160c)) + static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", [AZX_DRIVER_PCH] = "HDA Intel PCH", @@ -1976,8 +1981,8 @@ static int azx_probe_continue(struct azx *chip) * display codec needs the power and it can be released after probe. */ if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - /* Baytral/Braswell controllers don't need this power */ - if (pci->device != 0x0f04 && pci->device != 0x2284) + /* HSW/BDW controllers need this power */ + if (CONTROLLER_IN_GPU(pci)) hda->need_i915_power = 1; err = snd_hdac_i915_init(bus); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 73c20c0..407978b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2339,7 +2339,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) * can cover the codec power request, and so need not set this flag. * For previous platforms, there is no such power well feature. */ - if (is_valleyview_plus(codec)) + if (is_valleyview_plus(codec) || is_skylake(codec)) codec->core.link_power_control = 1; if (is_haswell_plus(codec) || is_valleyview_plus(codec)) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: add config to enable/disable asoc compress
by Jie Yang 03 Jun '15

03 Jun '15

From: "vivian,zhang" <vivian.zhang(a)intel.com> To reduce memory footprint, we can disable asoc compress when it is not used. Here add SND_SOC_COMPRESS to configure it. Signed-off-by: vivian,zhang <vivian.zhang(a)intel.com> Modified-by: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/soc.h | 7 +++++++ sound/soc/Kconfig | 10 +++++++++- sound/soc/Makefile | 6 +++++- 3 files changed, 21 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f6226914..6e2da70 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -430,7 +430,14 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +#ifdef CONFIG_SND_SOC_COMPRESS int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#else +static inline int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +{ + return -EPERM; +} +#endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 3ba52da..7e7a8e1 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,7 +9,7 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER - select SND_COMPRESS_OFFLOAD + select SND_COMPRESS_OFFLOAD if SND_SOC_COMPRESS ---help--- If you want ASoC support, you should say Y here and also to the @@ -30,6 +30,14 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_COMPRESS + bool "Asoc compress offload support" if EXPERT + default y + help + Say 'N" to disable Asoc compress offload support, which may reduce code + size about 10KB on X86_64 platform. + If unsure say Y. + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 974ba70..0832c3b 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o @@ -9,6 +9,10 @@ ifneq ($(CONFIG_SND_SOC_AC97_BUS),) snd-soc-core-objs += soc-ac97.o endif +ifneq ($(CONFIG_SND_SOC_COMPRESS),) +snd-soc-core-objs += soc-compress.o +endif + obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ -- 1.9.1

4 3

[alsa-devel] [PATCH 0/4] new fix for broadwell module removing failed issue
by Jie Yang 03 Jun '15

03 Jun '15

This patch series are based on broonie/sound.git for-next branch, commit 01a4d5083562fcdeab99d42b0873aabc5fcca018. The broadwell platform driver had module removing failed issue, and the two commits("ASoC: Intel: remove unused function hsw_pcm_free_modules()" and "ASoC: Intel: fix broadwell module removing failed issue") workaround it. But at the same time, we shouldn't leave the device as suspended state after module freed, it is not good to do runtime suspend at driver free either, so the fix was workaround only and not good enough. Here revert the old fix, and replace it with new unified procedure for runtime modules freeing: suspends firmware ==> frees runtime modules ==> unloads firmware. With this new procedure, we remove the runtime modules at both runtime suspend(D0->D3) and module(snd_soc_sst_haswell_pcm) removing, but make sure we do it only once. Jie Yang (4): Revert "ASoC: Intel: remove unused function hsw_pcm_free_modules()" Revert "ASoC: Intel: fix broadwell module removing failed issue" ASoC: Intel: check and clear runtime module pointer ASoC: Intel: handle haswell pcm suspend including runtime modules freeing sound/soc/intel/haswell/sst-haswell-pcm.c | 46 +++++++++++++++++++++++++------ 1 file changed, 37 insertions(+), 9 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH] ASoC: wm8731: initialize the hardware when loading the codec driver; use regmap for I/O
by Sergey Kiselev 02 Jun '15

02 Jun '15

Hi, This patch moves the request supplies and 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. The I/O is converted to regmap_write and regmap_update_bits. (Mostly as a side effect, since snd_soc_write and friends won't work before snd_soc_register_codec is called) Signed-off-by: Sergey Kiselev <sergey.kiselev(a)intel.com> --- wm8731.c | 107 ++++++++++++++++++++++++++++++++++++++++++--------------------- 1 file changed, 72 insertions(+), 35 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2245b6a..05fa4d6 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"}; @@ -118,7 +118,7 @@ static int wm8731_set_deemph(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Set deemphasis %d (%dHz)\n", best, wm8731_deemph[best]); - return snd_soc_update_bits(codec, WM8731_APDIGI, 0x6, val); + return regmap_update_bits(wm8731->regmap, WM8731_APDIGI, 0x6, val); } static int wm8731_get_deemph(struct snd_kcontrol *kcontrol, @@ -344,7 +344,7 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - u16 iface = snd_soc_read(codec, WM8731_IFACE) & 0xfff3; + unsigned int format = 0; int i = get_coeff(wm8731->sysclk, params_rate(params)); u16 srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1) | coeff_div[i].usb; @@ -358,28 +358,27 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, case 16: break; case 20: - iface |= 0x0004; + format = 0x4; break; case 24: - iface |= 0x0008; + format = 0x8; break; } wm8731_set_deemph(codec); - snd_soc_write(codec, WM8731_IFACE, iface); - return 0; + return regmap_update_bits(wm8731->regmap, WM8731_IFACE, 0x0c, format); } static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, WM8731_APDIGI) & 0xfff7; + struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); if (mute) - snd_soc_write(codec, WM8731_APDIGI, mute_reg | 0x8); + regmap_update_bits(wm8731->regmap, WM8731_APDIGI, 0x8, 0x8); else - snd_soc_write(codec, WM8731_APDIGI, mute_reg); + regmap_update_bits(wm8731->regmap, WM8731_APDIGI, 0x8, 0); return 0; } @@ -431,6 +430,7 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); u16 iface = 0; /* set master/slave audio interface */ @@ -482,7 +482,7 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - snd_soc_write(codec, WM8731_IFACE, iface); + regmap_write(wm8731->regmap, WM8731_IFACE, iface); return 0; } @@ -491,7 +491,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, { struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret; - u16 reg; switch (level) { case SND_SOC_BIAS_ON: @@ -511,13 +510,12 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, } /* Clear PWROFF, gate CLKOUT, everything else as-is */ - reg = snd_soc_read(codec, WM8731_PWR) & 0xff7f; - snd_soc_write(codec, WM8731_PWR, reg | 0x0040); + regmap_update_bits(wm8731->regmap, WM8731_PWR, 0xc0, 0x40); break; case SND_SOC_BIAS_OFF: if (wm8731->mclk) clk_disable_unprepare(wm8731->mclk); - snd_soc_write(codec, WM8731_PWR, 0xffff); + regmap_write(wm8731->regmap, WM8731_PWR, 0xff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); regcache_mark_dirty(wm8731->regmap); @@ -573,42 +571,53 @@ static struct snd_soc_dai_driver wm8731_dai = { static int wm8731_probe(struct snd_soc_codec *codec) { - struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - int ret = 0, i; + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int wm8731_request_supplies(struct device *dev, + struct wm8731_priv *wm8731) +{ + 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; } - wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* 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); + /* Clear POWEROFF, keep everything else disabled */ + regmap_write(wm8731->regmap, WM8731_PWR, 0x7f); /* 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); @@ -624,9 +633,7 @@ err_regulator_enable: /* 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); + wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -690,6 +697,13 @@ 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 +712,10 @@ 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 +729,12 @@ 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 +776,13 @@ 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 +791,10 @@ 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 +808,12 @@ 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; } -- Sergey

3 3

[alsa-devel] [PATCH v8 1/4] ASoC: arizona: Export functions to control subsystem DVFS
by Richard Fitzgerald 02 Jun '15

02 Jun '15

The WM5102 and WM8997 codecs have an internal dynamic clock booster. When this booster is active, the DCVDD voltage must be increased. If all the currently active audio paths can run with the root SYSCLK we can disable the booster, allowing us to turn down DCVDD voltage to save power. Previously this was being done by having the booster enable bit set as a side-effect of the LDO1 regulator driver, which is unexpected behaviour of a regulator and not compatible with using an external regulator. This patch exports functions to handle the booster enable and DCVDD voltage, with each relevant subsystem flagging whether it can currently run without the booster. Note that these subsystems are stateless and none of them are nestable, so there's no need for reference counting, we only need a simple boolean for each subsystem of whether their current condition could require the booster or will allow us to turn the codec down to lower operating power. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 128 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 13 +++++ sound/soc/codecs/wm5102.c | 12 +++-- sound/soc/codecs/wm8997.c | 11 +++- 4 files changed, 157 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index eff4b4d..b2d8b04 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -851,6 +851,134 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(arizona_hp_ev); +static int arizona_dvfs_enable(struct snd_soc_codec *codec) +{ + const struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int ret; + + ret = regulator_set_voltage(arizona->dcvdd, 1800000, 1800000); + if (ret) { + dev_err(codec->dev, "Failed to boost DCVDD: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(arizona->regmap, + ARIZONA_DYNAMIC_FREQUENCY_SCALING_1, + ARIZONA_SUBSYS_MAX_FREQ, + ARIZONA_SUBSYS_MAX_FREQ); + if (ret) { + dev_err(codec->dev, "Failed to enable subsys max: %d\n", ret); + regulator_set_voltage(arizona->dcvdd, 1200000, 1800000); + return ret; + } + + return 0; +} + +static int arizona_dvfs_disable(struct snd_soc_codec *codec) +{ + const struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int ret; + + ret = regmap_update_bits(arizona->regmap, + ARIZONA_DYNAMIC_FREQUENCY_SCALING_1, + ARIZONA_SUBSYS_MAX_FREQ, 0); + if (ret) { + dev_err(codec->dev, "Failed to disable subsys max: %d\n", ret); + return ret; + } + + ret = regulator_set_voltage(arizona->dcvdd, 1200000, 1800000); + if (ret) { + dev_err(codec->dev, "Failed to unboost DCVDD: %d\n", ret); + return ret; + } + + return 0; +} + +int arizona_dvfs_up(struct snd_soc_codec *codec, unsigned int flags) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + mutex_lock(&priv->dvfs_lock); + + if (!priv->dvfs_cached && !priv->dvfs_reqs) { + ret = arizona_dvfs_enable(codec); + if (ret) + goto err; + } + + priv->dvfs_reqs |= flags; +err: + mutex_unlock(&priv->dvfs_lock); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_dvfs_up); + +int arizona_dvfs_down(struct snd_soc_codec *codec, unsigned int flags) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int old_reqs; + int ret = 0; + + mutex_lock(&priv->dvfs_lock); + + old_reqs = priv->dvfs_reqs; + priv->dvfs_reqs &= ~flags; + + if (!priv->dvfs_cached && old_reqs && !priv->dvfs_reqs) + ret = arizona_dvfs_disable(codec); + + mutex_unlock(&priv->dvfs_lock); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_dvfs_down); + +int arizona_dvfs_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + mutex_lock(&priv->dvfs_lock); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (priv->dvfs_reqs) + ret = arizona_dvfs_enable(codec); + + priv->dvfs_cached = false; + break; + case SND_SOC_DAPM_PRE_PMD: + /* We must ensure DVFS is disabled before the codec goes into + * suspend so that we are never in an illegal state of DVFS + * enabled without enough DCVDD + */ + priv->dvfs_cached = true; + + if (priv->dvfs_reqs) + ret = arizona_dvfs_disable(codec); + break; + default: + break; + } + + mutex_unlock(&priv->dvfs_lock); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_dvfs_sysclk_ev); + +void arizona_init_dvfs(struct arizona_priv *priv) +{ + mutex_init(&priv->dvfs_lock); +} +EXPORT_SYMBOL_GPL(arizona_init_dvfs); + static unsigned int arizona_sysclk_48k_rates[] = { 6144000, 12288000, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 11ff899..e82ca21 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -60,6 +60,9 @@ #define ARIZONA_MAX_DAI 6 #define ARIZONA_MAX_ADSP 4 +#define ARIZONA_DVFS_SR1_RQ 0x001 +#define ARIZONA_DVFS_ADSP1_RQ 0x100 + struct arizona; struct wm_adsp; @@ -84,6 +87,10 @@ struct arizona_priv { unsigned int spk_ena:2; unsigned int spk_ena_pending:1; + + unsigned int dvfs_reqs; + struct mutex dvfs_lock; + bool dvfs_cached; }; #define ARIZONA_NUM_MIXER_INPUTS 103 @@ -245,6 +252,12 @@ struct arizona_fll { char clock_ok_name[ARIZONA_FLL_NAME_LEN]; }; +extern int arizona_dvfs_up(struct snd_soc_codec *codec, unsigned int flags); +extern int arizona_dvfs_down(struct snd_soc_codec *codec, unsigned int flags); +extern int arizona_dvfs_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); +extern void arizona_init_dvfs(struct arizona_priv *priv); + extern int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, int ok_irq, struct arizona_fll *fll); extern int arizona_set_fll_refclk(struct arizona_fll *fll, int source, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 0c6d1bc..b73e3a3 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -605,12 +605,13 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, regmap_write_async(regmap, patch[i].reg, patch[i].def); break; - - default: + case SND_SOC_DAPM_PRE_PMD: break; + default: + return 0; } - return 0; + return arizona_dvfs_sysclk_ev(w, kcontrol, event); } static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, @@ -1036,7 +1037,8 @@ static const struct snd_kcontrol_new wm5102_aec_loopback_mux = static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, wm5102_sysclk_ev, SND_SOC_DAPM_POST_PMU), + 0, wm5102_sysclk_ev, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, @@ -1909,6 +1911,8 @@ static int wm5102_probe(struct platform_device *pdev) wm5102->core.arizona = arizona; wm5102->core.num_inputs = 6; + arizona_init_dvfs(&wm5102->core); + wm5102->core.adsp[0].part = "wm5102"; wm5102->core.adsp[0].num = 1; wm5102->core.adsp[0].type = WMFW_ADSP2; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index a4d1177..2a129dc 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -106,11 +106,13 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, regmap_write_async(regmap, patch[i].reg, patch[i].def); break; - default: + case SND_SOC_DAPM_PRE_PMD: break; + default: + return 0; } - return 0; + return arizona_dvfs_sysclk_ev(w, kcontrol, event); } static const char *wm8997_osr_text[] = { @@ -409,7 +411,8 @@ static const struct snd_kcontrol_new wm8997_aec_loopback_mux = static const struct snd_soc_dapm_widget wm8997_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, wm8997_sysclk_ev, SND_SOC_DAPM_POST_PMU), + 0, wm8997_sysclk_ev, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, @@ -1126,6 +1129,8 @@ static int wm8997_probe(struct platform_device *pdev) wm8997->core.arizona = arizona; wm8997->core.num_inputs = 4; + arizona_init_dvfs(&wm8997->core); + for (i = 0; i < ARRAY_SIZE(wm8997->fll); i++) wm8997->fll[i].vco_mult = 1; -- 1.7.2.5

2 4

[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller
by Jyri Sarha 02 Jun '15

02 Jun '15

Find the configured DMA controller by asking for a DMA channel in the probe phase and releasing it right after. The controller device can be found via the dma_chan struct and the controller is recognized from the compatible property of its device node. The patch assumes EDMA if there is no device node. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Changes since first patch version: - Change enum names from MCASP_?DMA to PCM_?DMA - Return PCM_EDMA if !mcasp->dev->of_node - Do not ignore possible error code returned by davinci_mcasp_get_dma_type() sound/soc/davinci/davinci-mcasp.c | 68 +++++++++++++++++++++++++++++++++------ 1 file changed, 58 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d793494..3b78c3e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1565,6 +1565,49 @@ static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp) return ret; } +enum { + PCM_EDMA, + PCM_SDMA, +}; +static const char *sdma_prefix = "ti,omap"; + +static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) +{ + struct dma_chan *chan; + const char *tmp; + int ret = PCM_EDMA; + + if (!mcasp->dev->of_node) + return PCM_EDMA; + + tmp = mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data; + chan = dma_request_slave_channel_reason(mcasp->dev, tmp); + if (IS_ERR(chan)) { + if (PTR_ERR(chan) != -EPROBE_DEFER) + dev_err(mcasp->dev, + "Can't verify DMA configuration (%ld)\n", + PTR_ERR(chan)); + return PTR_ERR(chan); + } + BUG_ON(!chan->device || !chan->device->dev); + + if (chan->device->dev->of_node) + ret = of_property_read_string(chan->device->dev->of_node, + "compatible", &tmp); + else + dev_dbg(mcasp->dev, "DMA controller has no of-node\n"); + + dma_release_channel(chan); + if (ret) + return ret; + + dev_dbg(mcasp->dev, "DMA controller compatible = \"%s\"\n", tmp); + if (!strncmp(tmp, sdma_prefix, strlen(sdma_prefix))) + return PCM_SDMA; + + return PCM_EDMA; +} + static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -1763,27 +1806,32 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret != 0) goto err; - switch (mcasp->version) { + ret = davinci_mcasp_get_dma_type(mcasp); + switch (ret) { + case PCM_EDMA: #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_EDMA_SOC)) - case MCASP_VERSION_1: - case MCASP_VERSION_2: - case MCASP_VERSION_3: ret = edma_pcm_platform_register(&pdev->dev); - break; +#else + dev_err(&pdev->dev, "Missing SND_EDMA_SOC\n"); + ret = -EINVAL; #endif + break; + case PCM_SDMA: #if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_OMAP_SOC)) - case MCASP_VERSION_4: ret = omap_pcm_platform_register(&pdev->dev); - break; +#else + dev_err(&pdev->dev, "Missing SND_SDMA_SOC\n"); + ret = -EINVAL; #endif + break; + case -EPROBE_DEFER: + break; default: - dev_err(&pdev->dev, "Invalid McASP version: %d\n", - mcasp->version); - ret = -EINVAL; + dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); break; } -- 1.9.1

2 2

[alsa-devel] [PATCH 30/98] hdspm.h: use __u8, __u32 and __u64 from linux/types.h
by Mikko Rapeli 02 Jun '15

02 Jun '15

Fixes userspace compilation errors like: sound/hdspm.h:43:2: error: unknown type name ‘uint32_t’ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/hdspm.h | 40 ++++++++++++++++++---------------------- 1 file changed, 18 insertions(+), 22 deletions(-) diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index 5737332..c4db6f5 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,11 +20,7 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ -#ifdef __KERNEL__ #include <linux/types.h> -#else -#include <stdint.h> -#endif /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 @@ -46,15 +42,15 @@ enum hdspm_speed { /* -------------------- IOCTL Peak/RMS Meters -------------------- */ struct hdspm_peak_rms { - uint32_t input_peaks[64]; - uint32_t playback_peaks[64]; - uint32_t output_peaks[64]; + __u32 input_peaks[64]; + __u32 playback_peaks[64]; + __u32 output_peaks[64]; - uint64_t input_rms[64]; - uint64_t playback_rms[64]; - uint64_t output_rms[64]; + __u64 input_rms[64]; + __u64 playback_rms[64]; + __u64 output_rms[64]; - uint8_t speed; /* enum {ss, ds, qs} */ + __u8 speed; /* enum {ss, ds, qs} */ int status2; }; @@ -155,21 +151,21 @@ enum hdspm_syncsource { }; struct hdspm_status { - uint8_t card_type; /* enum hdspm_io_type */ + __u8 card_type; /* enum hdspm_io_type */ enum hdspm_syncsource autosync_source; - uint64_t card_clock; - uint32_t master_period; + __u64 card_clock; + __u32 master_period; union { struct { - uint8_t sync_wc; /* enum hdspm_sync */ - uint8_t sync_madi; /* enum hdspm_sync */ - uint8_t sync_tco; /* enum hdspm_sync */ - uint8_t sync_in; /* enum hdspm_sync */ - uint8_t madi_input; /* enum hdspm_madi_input */ - uint8_t channel_format; /* enum hdspm_madi_channel_format */ - uint8_t frame_format; /* enum hdspm_madi_frame_format */ + __u8 sync_wc; /* enum hdspm_sync */ + __u8 sync_madi; /* enum hdspm_sync */ + __u8 sync_tco; /* enum hdspm_sync */ + __u8 sync_in; /* enum hdspm_sync */ + __u8 madi_input; /* enum hdspm_madi_input */ + __u8 channel_format; /* enum hdspm_madi_channel_format */ + __u8 frame_format; /* enum hdspm_madi_frame_format */ } madi; } card_specific; }; @@ -184,7 +180,7 @@ struct hdspm_status { #define HDSPM_ADDON_TCO 1 struct hdspm_version { - uint8_t card_type; /* enum hdspm_io_type */ + __u8 card_type; /* enum hdspm_io_type */ char cardname[20]; unsigned int serial; unsigned short firmware_rev; -- 2.1.4

2 2

[alsa-devel] [PATCH v2 0/9] asoc: Add audio for sti platforms
by Arnaud Pouliquen 02 Jun '15

02 Jun '15

version 2: corrections after first review - bindings: . suppress useless dts declarations - uniperif_player.c: . Suppress standby mode in uniplayer ( will be part of a dedicated patch) . Deliver IEC and clock adjustement controls in separate patch - uniperif_reader.c: . kept as separate file as behavior and feature not same than uniplayer . minor corrections - codec/sti-sas.c . implement DAPM for DAC . rework registers declaration to separate device and driver part version 1: - First patches submission Arnaud Pouliquen (9): ASoC: sti: add binding for ASoc driver ASoC: sti: Add uniperipheral header file ASoC: sti: Add CPU DAI driver for playback ASoC: sti: Add CPU DAI driver for capture ASoC: sti: Add platform driver ASoC: Add ability to build sti drivers ASoC: Codec: Add sti platform codec ASoC: sti: Add clock adjustement control ASoC: sti: Add IEC control .../devicetree/bindings/sound/st,sti-asoc-card.txt | 158 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sti-sas.c | 645 +++++++++++ sound/soc/sti/Kconfig | 17 + sound/soc/sti/Makefile | 5 + sound/soc/sti/sti_platform.c | 642 ++++++++++ sound/soc/sti/uniperif.h | 1224 ++++++++++++++++++++ sound/soc/sti/uniperif_player.c | 1218 +++++++++++++++++++ sound/soc/sti/uniperif_reader.c | 459 ++++++++ 12 files changed, 4376 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt create mode 100644 sound/soc/codecs/sti-sas.c create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile create mode 100644 sound/soc/sti/sti_platform.c create mode 100644 sound/soc/sti/uniperif.h create mode 100644 sound/soc/sti/uniperif_player.c create mode 100644 sound/soc/sti/uniperif_reader.c -- 1.9.1

3 24

[alsa-devel] [PATCH 18/98] include/uapi/sound/emu10k1.h: added EMU10K1 version of DECLARE_BITMAP etc macros
by Mikko Rapeli 02 Jun '15

02 Jun '15

Fixes userspace compilation error: error: expected specifier-qualifier-list before ‘DECLARE_BITMAP’ DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ DECLARE_BITMAP etc macros are not meant for userspace headers and thus added here as private copies for emu10k.h. Fix was suggested by Arnd Bergmann <arnd(a)arndb.de> in message <2168807.4Yxh5gl11Q@wuerfel> on lkml. Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/emu10k1.h | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index ec1535b..7575f0f 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -34,6 +34,23 @@ #define EMU10K1_FX8010_PCM_COUNT 8 +/* + * Following definitions are copies from kernel headers to support compiling + * this header file in userspace. The definitions are not generally available + * in uapi headers so the needed things are copied here wtih __EMU10k1 prefix. + */ + +/* From include/linux/bitops.h */ +#define __EMU10K1_BITS_PER_BYTE 8 +/* From include/linux/kernel.h */ +#define __EMU10K1_DIV_ROUND_UP(n,d) (((n) + (d) - 1) / (d)) +/* From include/linux/bitops.h */ +#define __EMU10K1_BITS_TO_LONGS(nr) \ + __EMU10K1_DIV_ROUND_UP(nr, __EMU10K1_BITS_PER_BYTE * sizeof(long)) +/* From include/linux/types.h */ +#define __EMU10K1_DECLARE_BITMAP(name,bits) \ + unsigned long name[__EMU10K1_BITS_TO_LONGS(bits)] + /* instruction set */ #define iMAC0 0x00 /* R = A + (X * Y >> 31) ; saturation */ #define iMAC1 0x01 /* R = A + (-X * Y >> 31) ; saturation */ @@ -300,7 +317,7 @@ struct snd_emu10k1_fx8010_control_old_gpr { struct snd_emu10k1_fx8010_code { char name[128]; - DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ + __EMU10K1_DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ __u32 __user *gpr_map; /* initializers */ unsigned int gpr_add_control_count; /* count of GPR controls to add/replace */ @@ -313,11 +330,11 @@ struct snd_emu10k1_fx8010_code { unsigned int gpr_list_control_total; /* total count of GPR controls */ struct snd_emu10k1_fx8010_control_gpr __user *gpr_list_controls; /* listed GPR controls */ - DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ + __EMU10K1_DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ __u32 __user *tram_data_map; /* data initializers */ __u32 __user *tram_addr_map; /* map initializers */ - DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ + __EMU10K1_DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ __u32 __user *code; /* one instruction - 64 bits */ }; -- 2.1.4

2 3

[alsa-devel] [WM5102/ARIZONA] spi driver needs to support ACPI
by Christian Hartmann 02 Jun '15

02 Jun '15

Hi list, I need ACPI support for the arizona-core / arizona-spi driver for the ACPI ID WM5102. This audio codec is build into the Lenovo Yoga 851F. This device is a 32Bit (UEFI) system http://support.lenovo.com/de/de/products/tablets/yoga-series/yoga-tablet-2-… I have tested with vanilla kernels from kernel.org 3.19.y ,4.0.y and master (from torvalds) currently its 4.1.0-rc3. As distribution I am working with Fedora / Fedlet was the initial install image. With all vanilla kernels I got at first an error message from spi32766: [ 0.296297] pxa2xx-spi 80860F0E:00: cs1 >= max 1 [ 0.296311] spi_master spi32766: failed to add SPI device WM510205:00 from ACPI so I have at first patched as suggested by broonie the arizona-core commit 48203a0fede057d6a8c5f6872c53326395350f45 Author: somebody anybody <somebody(a)foobar.com> Date: Mon May 11 13:35:43 2015 +0200 spi-pxa2xx : raise chipselect Signed-off-by: somebody anybody <somebody(a)foobar.com> diff --git a/drivers/spi/spi-pxa2xx.c b/drivers/spi/spi-pxa2xx.c index e3223ac..9c919ea 100644 --- a/drivers/spi/spi-pxa2xx.c +++ b/drivers/spi/spi-pxa2xx.c @@ -1279,7 +1279,7 @@ pxa2xx_spi_acpi_get_pdata(struct platform_device *pdev) if (adev->pnp.unique_id && !kstrtoint(adev->pnp.unique_id, 0, &devid)) ssp->port_id = devid; - pdata->num_chipselect = 1; + pdata->num_chipselect = 20; pdata->enable_dma = true; return pdata; This leds the spi master do the further ACPI configuration and inital setup for the arizona-spi device, but adding the ACPI ID to it does not work yet as expected: The current last patchset on top of master is Author: somebody anybody <somebody(a)foobar.com> Date: Wed May 13 11:36:25 2015 +0200 arizona : added ACPI id of wm5102 to SPI driver, too Signed-off-by: somebody anybody <somebody(a)foobar.com> diff --git a/drivers/mfd/arizona-spi.c b/drivers/mfd/arizona-spi.c index 1e845f6..c19a4c1 100644 --- a/drivers/mfd/arizona-spi.c +++ b/drivers/mfd/arizona-spi.c @@ -17,6 +17,7 @@ #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/spi/spi.h> +#include <linux/acpi.h> #include <linux/of.h> #include <linux/mfd/arizona/core.h> @@ -82,6 +83,15 @@ static int arizona_spi_remove(struct spi_device *spi) return 0; } +#ifdef CONFIG_ACPI +static struct acpi_device_id wm5102_acpi_match[] = { + { "WM5102", 0}, + { "WM510205", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, wm5102_acpi_match); +#endif + static const struct spi_device_id arizona_spi_ids[] = { { "wm5102", WM5102 }, { "wm5110", WM5110 }, @@ -96,6 +106,9 @@ static struct spi_driver arizona_spi_driver = { .owner = THIS_MODULE, .pm = &arizona_pm_ops, .of_match_table = of_match_ptr(arizona_of_match), +#ifdef CONFIG_ACPI + .acpi_match_table = ACPI_PTR(wm5102_acpi_match), +#endif }, .probe = arizona_spi_probe, .remove = arizona_spi_remove, commit 268e21d29c942205e40f749c6d71c77839dc67da Author: somebody anybody <somebody(a)foobar.com> Date: Wed May 13 11:36:04 2015 +0200 arizona : added ACPI id of wm5102 to i2c driver Signed-off-by: somebody anybody <somebody(a)foobar.com> diff --git a/drivers/mfd/arizona-i2c.c b/drivers/mfd/arizona-i2c.c index ff782a5..45ece1b 100644 --- a/drivers/mfd/arizona-i2c.c +++ b/drivers/mfd/arizona-i2c.c @@ -18,6 +18,7 @@ #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/of.h> +#include <linux/acpi.h> #include <linux/mfd/arizona/core.h> @@ -85,6 +86,15 @@ static int arizona_i2c_remove(struct i2c_client *i2c) return 0; } +#ifdef CONFIG_ACPI +static struct acpi_device_id wm5102_acpi_match[] = { + { "WM5102", 0}, + { "WM510205", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, wm5102_acpi_match); +#endif + static const struct i2c_device_id arizona_i2c_id[] = { { "wm5102", WM5102 }, { "wm5110", WM5110 }, @@ -100,6 +110,9 @@ static struct i2c_driver arizona_i2c_driver = { .owner = THIS_MODULE, .pm = &arizona_pm_ops, .of_match_table = of_match_ptr(arizona_of_match), +#ifdef CONFIG_ACPI + .acpi_match_table = ACPI_PTR(wm5102_acpi_match), +#endif }, .probe = arizona_i2c_probe, .remove = arizona_i2c_remove, with these two patches the local master branch compiles fine, but at initialzing the arizona-spi driver gots a NULL pointer exception. I attach the whole dmesg of 4.1.0-rc3 (build 4.1.0.24-rc3). I need some assistance or hints to the right direction, I am looking myself and I also try what I can, but I think it would be easy peasy with some experts like here on this list :) PS: the message : spi_master spi32766: raised num_chipselect to 20 for WM510205:00 is a dev_err() line added to the function static acpi_status acpi_spi_add_device(), there I had raised the num_chipselect first (that was wrong, the raise is now done as in the first patch above. this is only a debug message and will be removed soon) PPS: for me it seems that there is a missing 'link' from WM5102 to the sst-acpi 80860F28:00 device, which complaints NOW / the first time with the three patches applied on top on master.... will have a look into it. I hope I got this device working with some more help from you. cheers chris

2 5

[alsa-devel] [PATCH] ALSA: hda/realtek - Add a fixup for another Acer Aspire 9420
by Takashi Iwai 02 Jun '15

02 Jun '15

Acer Aspire 9420 with ALC883 (1025:0107) needs the fixup for EAPD to make the sound working like other Aspire models. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94111 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 80be33ce1a9e..0320cb523d9e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2168,6 +2168,7 @@ static const struct hda_fixup alc882_fixups[] = { static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_FIXUP_ACER_EAPD), SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_FIXUP_ACER_EAPD), + SND_PCI_QUIRK(0x1025, 0x0107, "Acer Aspire", ALC883_FIXUP_ACER_EAPD), SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_FIXUP_ACER_EAPD), SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_FIXUP_ACER_EAPD), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_FIXUP_ACER_EAPD), -- 2.4.2

1 0

[alsa-devel] [PATCH] ASoC: DAPM: fix snd_soc_dapm_new_control() implicit declaration
by Vladimir Zapolskiy 02 Jun '15

02 Jun '15

The change fixes the following compilation problem: sound/soc/soc-dapm.c: In function 'dapm_kcontrol_data_alloc': sound/soc/soc-dapm.c:388:4: error: implicit declaration of function 'snd_soc_dapm_new_control' [-Werror=implicit-function-declaration] data->widget = snd_soc_dapm_new_control(widget->dapm, ^ sound/soc/soc-dapm.c:387:17: warning: assignment makes pointer from integer without a cast [enabled by default] data->widget = snd_soc_dapm_new_control(widget->dapm, ^ sound/soc/soc-dapm.c: At top level: sound/soc/soc-dapm.c:3269:1: error: conflicting types for 'snd_soc_dapm_new_control' snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, ^ In addition to the fix add static qualifier to snd_soc_dapm_new_control() function to silence checkpatch. Fixes: 02aa78abec ("ASoC: DAPM: Add APIs to create individual DAPM controls.") Signed-off-by: Vladimir Zapolskiy <vz(a)mleia.com> --- sound/soc/soc-dapm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 78d96e1..0d800ac 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -52,6 +52,11 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, const char *control, int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink)); + +static struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget); + struct snd_soc_dapm_widget * snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget); @@ -3265,7 +3270,7 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); -struct snd_soc_dapm_widget * +static struct snd_soc_dapm_widget * snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { -- 2.1.4

2 1

[alsa-devel] [PATCH v4 0/7] ASoC: intel - add skylake PCM driver
by Vinod Koul 02 Jun '15

02 Jun '15

SKL has HDA controller based audio subsystem with DSP and support for I2S, HDA, PDM links. The hda core code has been moved to sound/hda/ by Takashi which current HDA drivers use and will also be used by ASoC SKL driver. The SKL platform driver will load and create the soc_hdac_bus which embeds the hdac_bus, same for hdac_device (hda codecs) and hdac_stream (pcms) This is on top of hdac code in Takashi's topic/hda This patch provides the match function for asoc type hda codecs and let's them get enumerated by hdac. The second patch in this series adds the controller specific soc code. Common parts are in hdac core with changes introduced as part of SKL controller in soc part. Then we add the rest of controller PCM driver code (still HDA) and last patch breaks the HDA streams to host and link which will allow insertion of DSP in between these links. The subsequent series will add IPC driver for SKL (using common IPC routines), then DSP topology handlers, DSP code with I2S support and then lastly when DFW is accepted then its handlers. This patch series adds the hda codec match functions followed by asoc hda controller routines, then SKL PCM driver and last decouples the controller for splitting the links Fixes in v4: Updates changelog in patch1 and few other patches Address Takashi's comment Address Marks comments Jeeja KP (7): ASoC: hda - add ASoC HDA codec match function ALSA: hda - add new HDA registers ASoC: hda - add asoc hda core bus, controller and stream helpers ASoC: intel - add Skylake HDA platform driver ASoC: intel - add Skylake HDA audio driver ASoC: intel - add makefile support for SKL driver ASoC: intel - adds support for decoupled mode in skl driver include/sound/hda_register.h | 88 ++++ include/sound/soc-hda-codec.h | 49 ++ include/sound/soc-hdaudio.h | 360 +++++++++++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/hda/Kconfig | 3 + sound/soc/hda/Makefile | 4 + sound/soc/hda/soc-hda-codec.c | 89 ++++ sound/soc/hda/soc-hdac-bus.c | 115 +++++ sound/soc/hda/soc-hdac-controller.c | 296 +++++++++++ sound/soc/hda/soc-hdac-stream.c | 409 +++++++++++++++ sound/soc/intel/Kconfig | 17 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/hda-skl-pcm.c | 937 ++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/hda-skl.c | 670 ++++++++++++++++++++++++ sound/soc/intel/skylake/hda-skl.h | 74 +++ 17 files changed, 3117 insertions(+) create mode 100644 include/sound/soc-hda-codec.h create mode 100644 include/sound/soc-hdaudio.h create mode 100644 sound/soc/hda/Kconfig create mode 100644 sound/soc/hda/Makefile create mode 100644 sound/soc/hda/soc-hda-codec.c create mode 100644 sound/soc/hda/soc-hdac-bus.c create mode 100644 sound/soc/hda/soc-hdac-controller.c create mode 100644 sound/soc/hda/soc-hdac-stream.c create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/hda-skl-pcm.c create mode 100644 sound/soc/intel/skylake/hda-skl.c create mode 100644 sound/soc/intel/skylake/hda-skl.h -- 1.9.1

3 56

[alsa-devel] ASoC: PATCH: wm8731 - reinitialize regmap cache after hardware reset
by Sergey Kiselev 02 Jun '15

02 Jun '15

Hi, I am working on a machine driver that uses wm8731 codec and I've noticed that it will not work after rmmod/modprobe of that machine driver. The issue is that wm8731_probe() resets the hardware, but it doesn't reinitialize the regmap cache to the default values. The patch below fixes this issue. I also skimmed through other codecs, and some of them might have similar issue (wm8510.c, wm8711.c, wm8750.c, wm8753.c, wm8940.c, wm8960.c, wm8971.c, wm8974.c, wm8988.c, wm8990.c) Another way to avoid this issue might be resetting the codec in the *_spi_probe()/*_i2c_probe(), so that it only gets reset once when codec is loading. Thanks in advance for any comments/suggestions on the patch. Thanks, Sergey Signed-off-by: Sergey Kiselev <skiselev(a)gmail.com> diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2245b6a..76af75b 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) +static int wm8731_reset(struct snd_soc_codec *codec); static const char *wm8731_input_select[] = {"Line In", "Mic"}; @@ -665,6 +665,21 @@ static const struct regmap_config wm8731_regmap = { .num_reg_defaults = ARRAY_SIZE(wm8731_reg_defaults), }; +static int wm8731_reset(struct snd_soc_codec *codec) +{ + int ret; + struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_write(codec, WM8731_RESET, 0); + + if (ret != 0) + return ret; + + regmap_reinit_cache(wm8731->regmap, &wm8731_regmap); + + return 0; +} + #if defined(CONFIG_SPI_MASTER) static int wm8731_spi_probe(struct spi_device *spi) {

4 3

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller
by Jyri Sarha 02 Jun '15

02 Jun '15

Find the configured DMA controller by asking for a DMA channel in the probe phase and releasing it right after. The controller device can be found via the dma_chan struct and the controller is recognized from the compatible property of its device node. The patch assumes EDMA if there is no device node. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 61 ++++++++++++++++++++++++++++++++------- 1 file changed, 51 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d793494..5bc2712 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1565,6 +1565,46 @@ static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp) return ret; } +enum { + MCASP_EDMA = 1, + MCASP_SDMA, +}; +static const char *sdma_prefix = "ti,omap"; + +static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) +{ + struct dma_chan *chan; + const char *tmp; + int ret = MCASP_EDMA; + + tmp = mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data; + chan = dma_request_slave_channel_reason(mcasp->dev, tmp); + if (IS_ERR(chan)) { + if (PTR_ERR(chan) != -EPROBE_DEFER) + dev_err(mcasp->dev, + "Can't verify DMA configuration (%ld)\n", + PTR_ERR(chan)); + return PTR_ERR(chan); + } + BUG_ON(!chan->device || !chan->device->dev); + + if (chan->device->dev->of_node) + ret = of_property_read_string(chan->device->dev->of_node, + "compatible", &tmp); + else + dev_dbg(mcasp->dev, "DMA controller has no of-node\n"); + + dma_release_channel(chan); + if (ret) + return ret; + + dev_dbg(mcasp->dev, "DMA controller compatible = \"%s\"\n", tmp); + if (!strncmp(tmp, sdma_prefix, strlen(sdma_prefix))) + return MCASP_SDMA; + + return MCASP_EDMA; +} + static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -1763,27 +1803,28 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret != 0) goto err; - switch (mcasp->version) { + ret = -EINVAL; + switch (davinci_mcasp_get_dma_type(mcasp)) { + case MCASP_EDMA: #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_EDMA_SOC)) - case MCASP_VERSION_1: - case MCASP_VERSION_2: - case MCASP_VERSION_3: ret = edma_pcm_platform_register(&pdev->dev); - break; +#else + dev_err(&pdev->dev, "Missing SND_EDMA_SOC\n"); #endif + break; + case MCASP_SDMA: #if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_OMAP_SOC)) - case MCASP_VERSION_4: ret = omap_pcm_platform_register(&pdev->dev); - break; +#else + dev_err(&pdev->dev, "Missing SND_SDMA_SOC\n"); #endif + break; default: - dev_err(&pdev->dev, "Invalid McASP version: %d\n", - mcasp->version); - ret = -EINVAL; + dev_err(&pdev->dev, "No DMA controller found\n"); break; } -- 1.9.1

3 6

[alsa-devel] [PATCH v4 00/18] new MFD LoCoMo driver set
by Dmitry Eremin-Solenikov 02 Jun '15

02 Jun '15

Sharp Zaurus SL-5500 and SL-5600 use special companion Gate Array. With this patchset I tried to modernise and restructure the LoCoMo drivers. * Use platform bus and mfd core code * Use GPIO API instead of custom locomo GPIO code * Use irqdomains to manage IRQs * Split mixed locomolcd driver to lcd and backlight parts * Implement IRQ handling in GPIO driver * Implement SPI driver used as a host for mmc_spi device to control SD cards Changes since v3: * Removed magic values in locomo irq handling code * Using mutex in locomokbd driver * Dropped empty locomokbd_remove * Small rearrangement of headers/comments/etc in core and input drivers The following changes since commit b787f68c36d49bb1d9236f403813641efa74a031: Linux 4.1-rc1 (2015-04-26 17:59:10 -0700) are available in the git repository at: git://git.infradead.org/users/dbaryshkov/zaurus.git locomo-v4 for you to fetch changes up to 7c8e8e05ca27a4ad82356eb57ad12a7ca39bfd95: ARM: drop old LoCoMo driver (2015-05-25 14:29:23 +0300) ---------------------------------------------------------------- Dmitry Eremin-Solenikov (18): mfd: add new driver for Sharp LoCoMo leds: port locomo leds driver to new locomo core input: convert LoCoMo keyboard driver to use new locomo core input: locomokbd: differentiate between two Enter keys input: make LoCoMo keyboard driver support both poodle and collie video: backlight: add new locomo backlight driver video: lcd: add LoCoMo LCD driver gpio: port LoCoMo gpio support from old driver gpio: locomo: implement per-pin irq handling spi: add locomo SPI driver i2c: add locomo i2c driver ARM: sa1100: make collie use new locomo drivers ARM: sa1100: don't preallocate IRQ space for locomo ASoC: pxa: poodle: make use of new locomo GPIO interface ARM: pxa: poodle: use new LoCoMo driver ARM: pxa: poodle: don't preallocate IRQ space for locomo video: backlight: drop old locomo bl/lcd driver ARM: drop old LoCoMo driver arch/arm/common/Kconfig | 3 - arch/arm/common/Makefile | 1 - arch/arm/common/locomo.c | 914 ----------------------------- arch/arm/include/asm/hardware/locomo.h | 221 ------- arch/arm/mach-pxa/Kconfig | 1 - arch/arm/mach-pxa/include/mach/poodle.h | 8 +- arch/arm/mach-pxa/poodle.c | 58 +- arch/arm/mach-sa1100/Kconfig | 1 - arch/arm/mach-sa1100/collie.c | 213 +++++-- arch/arm/mach-sa1100/include/mach/collie.h | 16 +- arch/arm/mach-sa1100/include/mach/irqs.h | 19 +- drivers/gpio/Kconfig | 13 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-locomo.c | 285 +++++++++ drivers/i2c/busses/Kconfig | 12 + drivers/i2c/busses/Makefile | 1 + drivers/i2c/busses/i2c-locomo.c | 135 +++++ drivers/input/keyboard/Kconfig | 2 +- drivers/input/keyboard/locomokbd.c | 262 ++++----- drivers/leds/Kconfig | 2 +- drivers/leds/leds-locomo.c | 117 ++-- drivers/mfd/Kconfig | 10 + drivers/mfd/Makefile | 1 + drivers/mfd/locomo.c | 336 +++++++++++ drivers/spi/Kconfig | 10 + drivers/spi/Makefile | 1 + drivers/spi/spi-locomo.c | 332 +++++++++++ drivers/video/backlight/Kconfig | 16 +- drivers/video/backlight/Makefile | 3 +- drivers/video/backlight/locomo_bl.c | 153 +++++ drivers/video/backlight/locomo_lcd.c | 285 +++++++++ drivers/video/backlight/locomolcd.c | 255 -------- include/linux/mfd/locomo.h | 169 ++++++ sound/soc/pxa/poodle.c | 52 +- 34 files changed, 2205 insertions(+), 1703 deletions(-) delete mode 100644 arch/arm/common/locomo.c delete mode 100644 arch/arm/include/asm/hardware/locomo.h create mode 100644 drivers/gpio/gpio-locomo.c create mode 100644 drivers/i2c/busses/i2c-locomo.c create mode 100644 drivers/mfd/locomo.c create mode 100644 drivers/spi/spi-locomo.c create mode 100644 drivers/video/backlight/locomo_bl.c create mode 100644 drivers/video/backlight/locomo_lcd.c delete mode 100644 drivers/video/backlight/locomolcd.c create mode 100644 include/linux/mfd/locomo.h

5 26

[alsa-devel] [PATCH] ASoC: Intel: don't need compress offload for broadwell
by Jie Yang 02 Jun '15

02 Jun '15

We don't need compress offload feature for broadwell broadwell machine, here remove the non exist dependency. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index ee03dbd..3a08f79 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -79,7 +79,6 @@ config SND_SOC_INTEL_BROADWELL_MACH depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \ I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL - select SND_COMPRESS_OFFLOAD select SND_SOC_RT286 help This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell -- 1.9.1

2 1

[alsa-devel] [PATCH v7 1/4] ASoC: arizona: Export functions to control subsystem DVFS
by Richard Fitzgerald 02 Jun '15

02 Jun '15

The WM5102 and WM8997 codecs have an internal dynamic clock booster. When this booster is active, the DCVDD voltage must be increased. If all the currently active audio paths can run with the root SYSCLK we can disable the booster, allowing us to turn down DCVDD voltage to save power. Previously this was being done by having the booster enable bit set as a side-effect of the LDO1 regulator driver, which is unexpected behaviour of a regulator and not compatible with using an external regulator. This patch exports functions to handle the booster enable and DCVDD voltage, with each relevant subsystem flagging whether it can currently run without the booster. Note that these subsystems are stateless and none of them are nestable, so there's no need for reference counting, we only need a simple boolean for each subsystem of whether their current condition could require the booster or will allow us to turn the codec down to lower operating power. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 127 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 13 +++++ sound/soc/codecs/wm5102.c | 12 +++-- sound/soc/codecs/wm8997.c | 11 +++- 4 files changed, 156 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index eff4b4d..69965ae 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -851,6 +851,133 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(arizona_hp_ev); +static int arizona_dvfs_enable(struct snd_soc_codec *codec) +{ + const struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int ret; + + ret = regulator_set_voltage(arizona->dcvdd, 1800000, 1800000); + if (ret) { + dev_err(codec->dev, "Failed to boost DCVDD: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(arizona->regmap, + ARIZONA_DYNAMIC_FREQUENCY_SCALING_1, + ARIZONA_SUBSYS_MAX_FREQ, + ARIZONA_SUBSYS_MAX_FREQ); + if (ret) { + dev_err(codec->dev, "Failed to enable subsys max: %d\n", ret); + regulator_set_voltage(arizona->dcvdd, 1200000, 1800000); + return ret; + } + + return 0; +} + +static int arizona_dvfs_disable(struct snd_soc_codec *codec) +{ + const struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int ret; + + ret = regmap_update_bits(arizona->regmap, + ARIZONA_DYNAMIC_FREQUENCY_SCALING_1, + ARIZONA_SUBSYS_MAX_FREQ, 0); + if (ret) { + dev_err(codec->dev, "Failed to disable subsys max: %d\n", ret); + return ret; + } + + ret = regulator_set_voltage(arizona->dcvdd, 1200000, 1800000); + if (ret) { + dev_err(codec->dev, "Failed to unboost DCVDD: %d\n", ret); + return ret; + } + + return 0; +} + +int arizona_dvfs_up(struct snd_soc_codec *codec, unsigned int flags) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + mutex_lock(&priv->dvfs_lock); + + if (!priv->dvfs_cached && !priv->dvfs_reqs) { + ret = arizona_dvfs_enable(codec); + if (ret) + goto err; + } + + priv->dvfs_reqs |= flags; +err: + mutex_unlock(&priv->dvfs_lock); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_dvfs_up); + +int arizona_dvfs_down(struct snd_soc_codec *codec, unsigned int flags) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int old_reqs = priv->dvfs_reqs; + int ret = 0; + + mutex_lock(&priv->dvfs_lock); + + priv->dvfs_reqs &= ~flags; + + if (!priv->dvfs_cached && old_reqs && !priv->dvfs_reqs) + ret = arizona_dvfs_disable(codec); + + mutex_unlock(&priv->dvfs_lock); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_dvfs_down); + +int arizona_dvfs_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + mutex_lock(&priv->dvfs_lock); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (priv->dvfs_reqs) + ret = arizona_dvfs_enable(codec); + + priv->dvfs_cached = false; + break; + case SND_SOC_DAPM_PRE_PMD: + /* We must ensure DVFS is disabled before the codec goes into + * suspend so that we are never in an illegal state of DVFS + * enabled without enough DCVDD + */ + priv->dvfs_cached = true; + + if (priv->dvfs_reqs) + ret = arizona_dvfs_disable(codec); + break; + default: + break; + } + + mutex_unlock(&priv->dvfs_lock); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_dvfs_sysclk_ev); + +void arizona_init_dvfs(struct arizona_priv *priv) +{ + mutex_init(&priv->dvfs_lock); +} +EXPORT_SYMBOL_GPL(arizona_init_dvfs); + static unsigned int arizona_sysclk_48k_rates[] = { 6144000, 12288000, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 11ff899..e82ca21 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -60,6 +60,9 @@ #define ARIZONA_MAX_DAI 6 #define ARIZONA_MAX_ADSP 4 +#define ARIZONA_DVFS_SR1_RQ 0x001 +#define ARIZONA_DVFS_ADSP1_RQ 0x100 + struct arizona; struct wm_adsp; @@ -84,6 +87,10 @@ struct arizona_priv { unsigned int spk_ena:2; unsigned int spk_ena_pending:1; + + unsigned int dvfs_reqs; + struct mutex dvfs_lock; + bool dvfs_cached; }; #define ARIZONA_NUM_MIXER_INPUTS 103 @@ -245,6 +252,12 @@ struct arizona_fll { char clock_ok_name[ARIZONA_FLL_NAME_LEN]; }; +extern int arizona_dvfs_up(struct snd_soc_codec *codec, unsigned int flags); +extern int arizona_dvfs_down(struct snd_soc_codec *codec, unsigned int flags); +extern int arizona_dvfs_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); +extern void arizona_init_dvfs(struct arizona_priv *priv); + extern int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, int ok_irq, struct arizona_fll *fll); extern int arizona_set_fll_refclk(struct arizona_fll *fll, int source, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 0c6d1bc..b73e3a3 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -605,12 +605,13 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, regmap_write_async(regmap, patch[i].reg, patch[i].def); break; - - default: + case SND_SOC_DAPM_PRE_PMD: break; + default: + return 0; } - return 0; + return arizona_dvfs_sysclk_ev(w, kcontrol, event); } static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, @@ -1036,7 +1037,8 @@ static const struct snd_kcontrol_new wm5102_aec_loopback_mux = static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, wm5102_sysclk_ev, SND_SOC_DAPM_POST_PMU), + 0, wm5102_sysclk_ev, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, @@ -1909,6 +1911,8 @@ static int wm5102_probe(struct platform_device *pdev) wm5102->core.arizona = arizona; wm5102->core.num_inputs = 6; + arizona_init_dvfs(&wm5102->core); + wm5102->core.adsp[0].part = "wm5102"; wm5102->core.adsp[0].num = 1; wm5102->core.adsp[0].type = WMFW_ADSP2; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index a4d1177..2a129dc 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -106,11 +106,13 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, regmap_write_async(regmap, patch[i].reg, patch[i].def); break; - default: + case SND_SOC_DAPM_PRE_PMD: break; + default: + return 0; } - return 0; + return arizona_dvfs_sysclk_ev(w, kcontrol, event); } static const char *wm8997_osr_text[] = { @@ -409,7 +411,8 @@ static const struct snd_kcontrol_new wm8997_aec_loopback_mux = static const struct snd_soc_dapm_widget wm8997_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, wm8997_sysclk_ev, SND_SOC_DAPM_POST_PMU), + 0, wm8997_sysclk_ev, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, @@ -1126,6 +1129,8 @@ static int wm8997_probe(struct platform_device *pdev) wm8997->core.arizona = arizona; wm8997->core.num_inputs = 4; + arizona_init_dvfs(&wm8997->core); + for (i = 0; i < ARRAY_SIZE(wm8997->fll); i++) wm8997->fll[i].vco_mult = 1; -- 1.7.2.5

2 7

[alsa-devel] [PATCH] ASoC: rsnd: Document r8a7778-specific binding
by Geert Uytterhoeven 02 Jun '15

02 Jun '15

Add the missing r8a7778-specific compatible value, which is already in use since v4.1-rc1. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 62ece4c59da759a4..b6b3a786855f275b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -5,6 +5,7 @@ Required properties: "renesas,rcar_sound-gen1" if generation1, and "renesas,rcar_sound-gen2" if generation2 Examples with soctypes are: + - "renesas,rcar_sound-r8a7778" (R-Car M1A) - "renesas,rcar_sound-r8a7790" (R-Car H2) - "renesas,rcar_sound-r8a7791" (R-Car M2-W) - reg : Should contain the register physical address. -- 1.9.1

3 2

[alsa-devel] [RFC][PATCH] x86: remove vmalloc.h from asm/io.h
by Stephen Rothwell 02 Jun '15

02 Jun '15

Nothing in asm/io.h uses anything from vmalloc.h, so remove the include and fix up the build problems in an allmodconfig (64 bit and 32 bit) build. This may be the place where x86 builds get vmalloc.h implicitly included and that tends to hide places where vmalloc() et al are added to files but the include of vmalloc.h is forgotten. Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Ingo Molnar <mingo(a)redhat.com> Cc: "H. Peter Anvin" <hpa(a)zytor.com> Cc: <x86(a)kernel.org> Cc: Konrad Rzeszutek Wilk <konrad.wilk(a)oracle.com> Cc: Boris Ostrovsky <boris.ostrovsky(a)oracle.com> Cc: David Vrabel <david.vrabel(a)citrix.com> Cc: Anton Vorontsov <anton(a)enomsg.org> Cc: Colin Cross <ccross(a)android.com> Cc: Kees Cook <keescook(a)chromium.org> Cc: Tony Luck <tony.luck(a)intel.com> Cc: "Rafael J. Wysocki" <rjw(a)rjwysocki.net> Cc: Len Brown <lenb(a)kernel.org> Cc: Kristen Carlson Accardi <kristen(a)linux.intel.com> Cc: Viresh Kumar <viresh.kumar(a)linaro.org> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: "K. Y. Srinivasan" <kys(a)microsoft.com> Cc: Haiyang Zhang <haiyangz(a)microsoft.com> Cc: Hiral Patel <hiralpat(a)cisco.com> Cc: Suma Ramars <sramars(a)cisco.com> Cc: Brian Uchino <buchino(a)cisco.com> Cc: "James E.J. Bottomley" <JBottomley(a)odin.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Andrew Morton <akpm(a)linux-foundation.org> Suggested-by: David Miller <davem(a)davemloft.net> Signed-off-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- Based in Linus' tree of today. There are probably more places that need vmalloc.h included, but this passes 64 bit and 32 bit allmodconfig builds, so is a place to start. Dave Miller suggested that I start this journey. arch/x86/include/asm/io.h | 2 -- arch/x86/kernel/crash.c | 1 + arch/x86/kernel/machine_kexec_64.c | 1 + arch/x86/mm/pageattr-test.c | 1 + arch/x86/mm/pageattr.c | 1 + arch/x86/xen/p2m.c | 1 + drivers/acpi/apei/erst.c | 1 + drivers/cpufreq/intel_pstate.c | 1 + drivers/dma/mic_x100_dma.c | 1 + drivers/net/hyperv/netvsc.c | 1 + drivers/net/hyperv/rndis_filter.c | 1 + drivers/scsi/fnic/fnic_debugfs.c | 1 + drivers/scsi/fnic/fnic_trace.c | 1 + sound/pci/asihpi/hpioctl.c | 1 + 14 files changed, 13 insertions(+), 2 deletions(-) diff --git a/arch/x86/include/asm/io.h b/arch/x86/include/asm/io.h index 34a5b93704d3..5791e7ace9db 100644 --- a/arch/x86/include/asm/io.h +++ b/arch/x86/include/asm/io.h @@ -197,8 +197,6 @@ extern void set_iounmap_nonlazy(void); #include <asm-generic/iomap.h> -#include <linux/vmalloc.h> - /* * Convert a virtual cached pointer to an uncached pointer */ diff --git a/arch/x86/kernel/crash.c b/arch/x86/kernel/crash.c index c76d3e37c6e1..e068d6683dba 100644 --- a/arch/x86/kernel/crash.c +++ b/arch/x86/kernel/crash.c @@ -22,6 +22,7 @@ #include <linux/elfcore.h> #include <linux/module.h> #include <linux/slab.h> +#include <linux/vmalloc.h> #include <asm/processor.h> #include <asm/hardirq.h> diff --git a/arch/x86/kernel/machine_kexec_64.c b/arch/x86/kernel/machine_kexec_64.c index 415480d3ea84..11546b462fa6 100644 --- a/arch/x86/kernel/machine_kexec_64.c +++ b/arch/x86/kernel/machine_kexec_64.c @@ -17,6 +17,7 @@ #include <linux/ftrace.h> #include <linux/io.h> #include <linux/suspend.h> +#include <linux/vmalloc.h> #include <asm/init.h> #include <asm/pgtable.h> diff --git a/arch/x86/mm/pageattr-test.c b/arch/x86/mm/pageattr-test.c index 6629f397b467..8ff686aa7e8c 100644 --- a/arch/x86/mm/pageattr-test.c +++ b/arch/x86/mm/pageattr-test.c @@ -9,6 +9,7 @@ #include <linux/random.h> #include <linux/kernel.h> #include <linux/mm.h> +#include <linux/vmalloc.h> #include <asm/cacheflush.h> #include <asm/pgtable.h> diff --git a/arch/x86/mm/pageattr.c b/arch/x86/mm/pageattr.c index 89af288ec674..bedfc794b4ba 100644 --- a/arch/x86/mm/pageattr.c +++ b/arch/x86/mm/pageattr.c @@ -14,6 +14,7 @@ #include <linux/percpu.h> #include <linux/gfp.h> #include <linux/pci.h> +#include <linux/vmalloc.h> #include <asm/e820.h> #include <asm/processor.h> diff --git a/arch/x86/xen/p2m.c b/arch/x86/xen/p2m.c index b47124d4cd67..8b7f18e200aa 100644 --- a/arch/x86/xen/p2m.c +++ b/arch/x86/xen/p2m.c @@ -67,6 +67,7 @@ #include <linux/seq_file.h> #include <linux/bootmem.h> #include <linux/slab.h> +#include <linux/vmalloc.h> #include <asm/cache.h> #include <asm/setup.h> diff --git a/drivers/acpi/apei/erst.c b/drivers/acpi/apei/erst.c index ed65e9c4b5b0..3670bbab57a3 100644 --- a/drivers/acpi/apei/erst.c +++ b/drivers/acpi/apei/erst.c @@ -35,6 +35,7 @@ #include <linux/nmi.h> #include <linux/hardirq.h> #include <linux/pstore.h> +#include <linux/vmalloc.h> #include <acpi/apei.h> #include "apei-internal.h" diff --git a/drivers/cpufreq/intel_pstate.c b/drivers/cpufreq/intel_pstate.c index 6414661ac1c4..2ba53f4f6af2 100644 --- a/drivers/cpufreq/intel_pstate.c +++ b/drivers/cpufreq/intel_pstate.c @@ -26,6 +26,7 @@ #include <linux/fs.h> #include <linux/debugfs.h> #include <linux/acpi.h> +#include <linux/vmalloc.h> #include <trace/events/power.h> #include <asm/div64.h> diff --git a/drivers/dma/mic_x100_dma.c b/drivers/dma/mic_x100_dma.c index 6de2e677be04..74d9db05a5ad 100644 --- a/drivers/dma/mic_x100_dma.c +++ b/drivers/dma/mic_x100_dma.c @@ -22,6 +22,7 @@ #include <linux/module.h> #include <linux/io.h> #include <linux/seq_file.h> +#include <linux/vmalloc.h> #include "mic_x100_dma.h" diff --git a/drivers/net/hyperv/netvsc.c b/drivers/net/hyperv/netvsc.c index ea091bc5ff09..1e09243d5449 100644 --- a/drivers/net/hyperv/netvsc.c +++ b/drivers/net/hyperv/netvsc.c @@ -28,6 +28,7 @@ #include <linux/slab.h> #include <linux/netdevice.h> #include <linux/if_ether.h> +#include <linux/vmalloc.h> #include <asm/sync_bitops.h> #include "hyperv_net.h" diff --git a/drivers/net/hyperv/rndis_filter.c b/drivers/net/hyperv/rndis_filter.c index 9118cea91882..35a482d526d9 100644 --- a/drivers/net/hyperv/rndis_filter.c +++ b/drivers/net/hyperv/rndis_filter.c @@ -27,6 +27,7 @@ #include <linux/netdevice.h> #include <linux/if_vlan.h> #include <linux/nls.h> +#include <linux/vmalloc.h> #include "hyperv_net.h" diff --git a/drivers/scsi/fnic/fnic_debugfs.c b/drivers/scsi/fnic/fnic_debugfs.c index 5980c10c734d..d6498fabe628 100644 --- a/drivers/scsi/fnic/fnic_debugfs.c +++ b/drivers/scsi/fnic/fnic_debugfs.c @@ -18,6 +18,7 @@ #include <linux/module.h> #include <linux/errno.h> #include <linux/debugfs.h> +#include <linux/vmalloc.h> #include "fnic.h" static struct dentry *fnic_trace_debugfs_root; diff --git a/drivers/scsi/fnic/fnic_trace.c b/drivers/scsi/fnic/fnic_trace.c index 65a9bde26974..4e15c4bf0795 100644 --- a/drivers/scsi/fnic/fnic_trace.c +++ b/drivers/scsi/fnic/fnic_trace.c @@ -21,6 +21,7 @@ #include <linux/spinlock.h> #include <linux/kallsyms.h> #include <linux/time.h> +#include <linux/vmalloc.h> #include "fnic_io.h" #include "fnic.h" diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 6610bd096fc9..d17937b92331 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -32,6 +32,7 @@ #include <linux/pci.h> #include <linux/stringify.h> #include <linux/module.h> +#include <linux/vmalloc.h> #ifdef MODULE_FIRMWARE MODULE_FIRMWARE("asihpi/dsp5000.bin"); -- 2.1.4 -- Cheers, Stephen Rothwell sfr(a)canb.auug.org.au

4 6

[alsa-devel] [PATCH 0/2] Update ALSA driver to use media controller API
by Shuah Khan 01 Jun '15

01 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. 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/card.c | 5 ++++ sound/usb/card.h | 12 +++++++++ sound/usb/pcm.c | 23 +++++++++++++++++- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 58 +++++++++++++++++++++++++++++++++++++++++++- sound/usb/quirks.h | 6 +++++ sound/usb/stream.c | 40 ++++++++++++++++++++++++++++++ sound/usb/usbaudio.h | 1 + 10 files changed, 179 insertions(+), 2 deletions(-) -- 2.1.4

3 7

[alsa-devel] [PATCH 01/47] ASoC: cs42l52: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 01 Jun '15

01 Jun '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all other manual access to codec->dapm with snd_soc_codec_get_dapm(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Series depends on topic/dapm --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 3c49a75..4de52c9 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -897,7 +897,7 @@ static int cs42l52_set_bias_level(struct snd_soc_codec *codec, CS42L52_PWRCTL1_PDN_CODEC, 0); break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { regcache_cache_only(cs42l52->regmap, false); regcache_sync(cs42l52->regmap); } @@ -955,7 +955,7 @@ static void cs42l52_beep_work(struct work_struct *work) struct cs42l52_private *cs42l52 = container_of(work, struct cs42l52_private, beep_work); struct snd_soc_codec *codec = cs42l52->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); int i; int val = 0; int best = 0; -- 2.1.4

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[alsa-devel] Override default behavior when headphones are connected upon user request on wm8962
by Santiago Zarate 01 Jun '15

01 Jun '15

Hello, Currently i'm trying to override the default behavior when headphones are plugged, We actually need to play audio under certain situations, and one of the use cases is when the headphones are connected, but a sound needs to be played over the speakers, so someone else is aware of what's happening. We're thinking about using a GPIO that it's available from our application or a simple module parameter, but the problem i'm facing right now is: 1: Where to specify that particular path or the behavior i need, i've been looking at [2], and adding a SOC_DAPM_PIN_SWITCH[1] but doesn't matter what the state is, if the headphones are on, i can't turn on the speakers to play sounds at the same time. i'm able to do it, by disabling the hp detection pin, and controlling the audio over udev events. so i'm pretty sure that it would be possible to do this, with the hp detection pin. which takes us to #2: 2: How can i trigger an event based on the change of a module parameter or a gpio port (Thinking both would be the same)... i've tried to alter hp_event <http://lxr.free-electrons.com/ident?v=3.16;i=hp_event> without any success... even returning 0 on any event and still nothing, which drives me to belive that i'm doing something completely wrong AND the codec/driver is among the last ones to get notified of the presence of the hp The idea is: if override port or parameter is set to true, trigger the audio override and play sound on Speaker and HP at the same time. right now i'm working with imx-wm8962 driver and the matching codec, http://mailman.alsa-project.org/pipermail/alsa-devel/2011-February/036370.h… [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2010-December/035055.h… [2] Thanks in advance! -- Santiago Zarate http://foursixnine.io “Dreams permit each and every one of us to be quietly and safely insane every night of our lives.” - William Dement

1 0

[alsa-devel] Audio play problem with tlv320aic3106 in mainline.
by Enric Balletbo Serra 01 Jun '15

01 Jun '15

Dear all, Recently I sent some patches to add support for a new board based on AM335x processor [1]. The board has the TLV320AIC3106 audio codec, the MCLK signal is provided by an external oscillator of 12MHz. The problem is that when I play a wav file for example I heard the song a double speed. This is with v4.0 and current mainline. In dts file I defined "ti,codec-clock-rate = <12000000>;" which is the correct value. Curiously when I changed the clock rate value to 24000000 I heard the song at normal speed. AFAIK the correct value for clock-rate is 12MHz so I suspect that something is wrong in the driver, maybe it sets a wrong clock divider/multiplier ? Anyone can point me in the right direction to fix the issue ? I see that am335x-evm and am335x-pepper boards also uses this audio codec, did anyone with those boards have the same problem? [1] https://patchwork.kernel.org/patch/6480071/ Thanks in advance, Enric

2 5

[alsa-devel] Remarks about alsa-lib-1.0.29/test/pcm*.c
by Christophe Lohr 01 Jun '15

01 Jun '15

Hello, I'm looking about examples in alsa-lib-1.0.29/test/ I have two questions about: pcm.c line 71: In case of a FLOAT format, maxval is 1.0, isn't it? pcm_min.c lin 79: Variable err does not refers to the previous call? Did I understood well? Best regards Christophe

2 1

[alsa-devel] [RFC] sound: ppc: keywest: check if DEQ was already instantiated
by Wolfram Sang 01 Jun '15

01 Jun '15

Due to changes in i2c-powermac, for some Macs the DEQ is instantiated now in i2c-powermac while some need the instantiation here in the keywest sound driver. The proper fix is to convert this driver to AOA. Until then support both ways of instantiation. Before this patch, some Macs lost sound because the sound driver failed when instantiating the DEQ for the second time. Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> --- Dan, Mark: I hope I found a generic but not too intrusive solution to the doubled instantiation problem. Can you please test this patch on top of this one "sound: ppc: keywest: drop using attach adapter" from last time? Check if you have sound and send the dmesg output, please. I hope it compiles even, couldn't test that this time. sound/ppc/keywest.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index d7627bae08362e..6120a067494a44 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -31,10 +31,15 @@ */ static struct pmac_keywest *keywest_ctx; +static bool keywest_probed; static int keywest_probe(struct i2c_client *client, const struct i2c_device_id *id) { + keywest_probed = true; + /* If instantiated via i2c-powermac, we still need to set the client */ + if (!keywest_ctx->client) + keywest_ctx->client = client; i2c_set_clientdata(client, keywest_ctx); return 0; } @@ -92,7 +97,8 @@ static int keywest_remove(struct i2c_client *client) static const struct i2c_device_id keywest_i2c_id[] = { - { "keywest", 0 }, + { "MAC,tas3004", 0 }, /* instantiated by i2c-powermac */ + { "keywest", 0 }, /* instantiated by us if needed */ { } }; @@ -149,8 +155,13 @@ int snd_pmac_keywest_init(struct pmac_keywest *i2c) return err; } + /* There was already a device from i2c-powermac. Great, let's return */ + if (keywest_probed) + return 0; + /* We assume Macs have consecutive I2C bus numbers starting at 0 */ while (adap) { + /* Scan for devices to be bound to */ err = keywest_attach_adapter(adap); if (!err) return 0; -- 2.1.4

3 7

[alsa-devel] [PATCH] ALSA: hda - Fix jack detection at resume with VT codecs
by Takashi Iwai 31 May '15

31 May '15

VT202x codecs seem requiring some delay after the resume D0 power transition for making the jack detection working again. Without the delay soon after D0, the jack is always detected as unplugged. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98921 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_via.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 31a95cca015d..bab6c04932aa 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -449,6 +449,15 @@ static int via_suspend(struct hda_codec *codec) return 0; } + +static int via_resume(struct hda_codec *codec) +{ + /* some delay here to make jack detection working (bko#98921) */ + msleep(10); + codec->patch_ops.init(codec); + regcache_sync(codec->core.regmap); + return 0; +} #endif #ifdef CONFIG_PM @@ -475,6 +484,7 @@ static const struct hda_codec_ops via_patch_ops = { .stream_pm = snd_hda_gen_stream_pm, #ifdef CONFIG_PM .suspend = via_suspend, + .resume = via_resume, .check_power_status = via_check_power_status, #endif }; -- 2.4.2

1 0

[alsa-devel] Realtek ALC892 / snd-hda-intel breakage between 3.16 & 3.19
by Sander Smeenk 31 May '15

31 May '15

Hi, After upgrading from 3.16.0-38-generic (Ubuntu Utopic) to 3.19.0-18-generic (Ubuntu Vivid) my audio stopped working. The issue appears to be in the snd-hda-intel driver. 'Alsa-info' is attached as gzip text, or online in paste bins: - 3.19, broken: https://8n1.org/10575/e5c1 - 3.16, working: https://8n1.org/10576/0631 With 3.19, tools fail like: | [sanders@msi:~] % aplay test.wav | Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo | ALSA lib pcm_pulse.c:752:(pulse_prepare) PulseAudio: Unable to create stream: No such entity | aplay: set_params:1297: Unable to install hw params: | [ .. ] | | [sanders@msi:~] % mplayer -ao alsa test.wav | Selected audio codec: Uncompressed PCM [pcm] | [ .. ] | AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: | 176400->176400) | [AO_ALSA] alsa-lib: pcm_pulse.c:752:(pulse_prepare) PulseAudio: Unable to create stream: No such entity | [AO_ALSA] Unable to set hw-parameters: Input/output error | Failed to initialize audio driver 'alsa' | Could not open/initialize audio device -> no sound. And i have only one card listed in /proc/asound/cards, index 1 not 0: | [sanders@msi:~] % cat /proc/asound/cards | 1 [PCH ]: HDA-Intel - HDA Intel PCH | HDA Intel PCH at 0xf7b10000 irq 36 This all works perfectly fine with 3.16. Please let me know when i can supply more information to dig into this! Regards, -Sndr. -- | Alles hat ein ende, nur die wurst hat zwei. | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2

3 4

[alsa-devel] [PATCH] ALSA: usb-audio: don't try to get Outlaw RR2150 sample rate
by Eric Wong 30 May '15

30 May '15

This quirk allows us to avoid the noisy: current rate 0 is different from the runtime rate message every time playback starts. While USB DAC in the RR2150 supports reading the sample rate, it never returns a sample rate other than zero in my observation with common sample rates. Signed-off-by: Eric Wong <normalperson(a)yhbt.net> Cc: Joe Turner <joe(a)oampo.co.uk> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 32631a8..2ea5b33 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1118,6 +1118,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema */ case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ + case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ return true; } return false; -- EW

2 1

[alsa-devel] [PATCH] ASoC: Intel: fix broadwell module removing failed issue
by Jie Yang 30 May '15

30 May '15

From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> In haswell-pcm module unloading, we can't free runtime modules directly, for they may be already freed in runtime suspend. Here add executing suspend call to unload runtime modules, only for status not equal to RPM_SUSPEND, to fix broadwell module removing failed issue. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 23ae040..bd96629 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -1118,8 +1118,10 @@ static int hsw_pcm_remove(struct snd_soc_platform *platform) snd_soc_platform_get_drvdata(platform); int i; + /* execute a suspend call to unload all FW resources */ + if (!pm_runtime_status_suspended(platform->dev)) + pm_runtime_put_sync_suspend(platform->dev); pm_runtime_disable(platform->dev); - hsw_pcm_free_modules(priv_data); for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { if (hsw_dais[i].playback.channels_min) -- 1.9.1

5 8

[alsa-devel] [PATCH - alsa-plugins/arcam-av ] arcam-av: Include sys/select.h for fd_set definition
by Khem Raj 30 May '15

30 May '15

fixes build errors on non-glibc based systems arcam-av/arcam_av.c:389:2: error: unknown type name 'fd_set' Signed-off-by: Khem Raj <raj.khem(a)gmail.com> --- arcam-av/arcam_av.c | 1 + 1 file changed, 1 insertion(+) diff --git a/arcam-av/arcam_av.c b/arcam-av/arcam_av.c index 0491fc6..fff96f9 100644 --- a/arcam-av/arcam_av.c +++ b/arcam-av/arcam_av.c @@ -31,6 +31,7 @@ #include <unistd.h> #include <sys/ipc.h> +#include <sys/select.h> #include <sys/shm.h> #include <sys/stat.h> #include <sys/stat.h> -- 2.1.4

2 1

[alsa-devel] [RFC][PATCH 0/2] ALSA: control: add dimen validator
by Takashi Sakamoto 30 May '15

30 May '15

Hi, This patchset adds a validator for dimen field in control element information. The dimen information is used to represent channel matrix of the control element, thus the number of elements in the matrix should be within the number of channels in the control element. Current implementation allows each drivers to set arbitrary number to the information. When the information is wrong, it can cause buffer-over-run to userspace applications depending on application implementation. Actually, the dimen information is scarcely used by current drivers and applications, while it's better for ALSA core to ensure the information is enough safe. Takashi Sakamoto (2): ALSA: control: add dimension validator for userspace element ALSA: control: add dimension validator for kernel driver sound/core/control.c | 62 ++++++++++++++++++++++++++++++++++++++-------------- 1 file changed, 46 insertions(+), 16 deletions(-) -- 2.1.4

2 5

[alsa-devel] [PATCH - alsa-plugins/arcam-av ] arcam-av: Include sys/select.h for fd_set definition
by Khem Raj 30 May '15

30 May '15

fixes build errors on non-glibc based systems arcam-av/arcam_av.c:389:2: error: unknown type name 'fd_set' Signed-off-by: Khem Raj <raj.khem(a)gmail.com> --- arcam-av/arcam_av.c | 1 + 1 file changed, 1 insertion(+) diff --git a/arcam-av/arcam_av.c b/arcam-av/arcam_av.c index 0491fc6..fff96f9 100644 --- a/arcam-av/arcam_av.c +++ b/arcam-av/arcam_av.c @@ -31,6 +31,7 @@ #include <unistd.h> #include <sys/ipc.h> +#include <sys/select.h> #include <sys/shm.h> #include <sys/stat.h> #include <sys/stat.h> -- 2.1.4

1 0

[alsa-devel] randconfig build error with next-20150529, in sound/soc/qcom/lpass-ipq806x
by Jim Davis 29 May '15

29 May '15

Building with the attached random configuration file, warning: (SND_SOC_STORM) selects SND_SOC_LPASS_IPQ806X which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && SND_SOC_QCOM) warning: (INTEL_SOC_DTS_IOSF_CORE && PUNIT_ATOM_DEBUG) selects IOSF_MBI which has unmet direct dependencies (PCI) ERROR: "asoc_qcom_lpass_cpu_dai_probe" [sound/soc/qcom/snd-soc-lpass-ipq806x.ko] undefined! ERROR: "asoc_qcom_lpass_cpu_platform_probe" [sound/soc/qcom/snd-soc-lpass-ipq806x.ko] undefined! ERROR: "asoc_qcom_lpass_cpu_platform_remove" [sound/soc/qcom/snd-soc-lpass-ipq806x.ko] undefined! ERROR: "asoc_qcom_lpass_cpu_dai_ops" [sound/soc/qcom/snd-soc-lpass-ipq806x.ko] undefined!

1 0

[alsa-devel] [PATCH v3 0/3] ASoC topology
by Liam Girdwood 29 May '15

29 May '15

ASoC topology (formerly called dynamic firmware) allows audio DSP topology to be defined as part of the firmware package and not hard coded into the driver. This allows DSP vendors to ship multiple different firmwares or firmwares with programmable topologies for different devices (i.e. phones, tablets, TVs) using a single DSP driver. The DSP topology can be contained as part of the firmware file or exist as separate files. The topology core and file format is generic text and not tied to any single vendor or device. The topology core consists of kernel and userspace components :- Kernel. The kernel core reads in the topology data and builds the driver capabilities and audio topology based on this data. This includes building the DAPM graph, setting PCM and DAI capabilities, registering kcontrols and registering DAPM widgets. The userspace tool allows the creation of binary DSP FW topology files based on a topology text files. The tool is temporarily available on github and its intended to be merged into the alsa-utils package after a clean up. https://github.com/lgirdwood/topology.git Changes since V2 :- o Moved TLV channel maps into TLV UAPI header. o Report error in logs when no handler exists for vendor data types. o Fix comments on kcontrol/widget ops to match core code. Changes since RFC :- o Fixed patch series to survive bisection builds. o Removed widget ID enum from UAPI. o Added channel mappings to controls. o Added TLV macro patch to UAPI. o Fixed locking for runtime and init time topology loading. o Moved the UAPI group index field to the block header. Changes since V1 (back in 2012) :- o Name changed to ASoC Topology o Structures from soc.h and soc-dapm.h no longer exposed as UAPIs o Moved to component model from codec/platform model. o Added reserved fields in each UAPI structure. o UAPI structures packed and endianess defined for each structure. o Added support for PCM, DAI and codec <-> codec link capabilities. o Added support for client drivers manifests. o Added support for private data for each dynamic object type. o Single list of dynamic objects rather than list of each object type. Liam

1 0

[alsa-devel] [PATCH] sound/oss: use schedule_timeout_interruptible()
by Nicholas Mc Guire 29 May '15

29 May '15

API consolidation with coccinelle found: ./sound/oss/msnd_pinnacle.c:1292:2-18: consolidation with schedule_timeout_*() recommended This is a 1:1 conversion of the current calls to an available helper only - so only an API consolidation to improve readability. Patch was compile tested with x86_64_defconfig Patch is against 4.1-rc5 (localversion-next is -next-20150529) Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- sound/oss/msnd_pinnacle.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index a8ceef8..a8bb4a0 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -1288,8 +1288,7 @@ static int __init calibrate_adc(WORD srate) & ~0x0001, dev.SMA + SMA_wCurrHostStatusFlags); if (msnd_send_word(&dev, 0, 0, HDEXAR_CAL_A_TO_D) == 0 && chk_send_dsp_cmd(&dev, HDEX_AUX_REQ) == 0) { - __set_current_state(TASK_INTERRUPTIBLE); - schedule_timeout(HZ / 3); + schedule_timeout_interruptible(HZ / 3); return 0; } printk(KERN_WARNING LOGNAME ": ADC calibration failed\n"); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ALSA: jack: Fix the id uniqueness check
by Takashi Iwai 29 May '15

29 May '15

snd_kctl_jack_new() tries to assign a unique index number when a name string that has been already registered is passed. However, it checks with the base string without "Jack" suffix, so it never hits. Fix the call with the properly processed name string instead. Fixes: b8dd086674cf 'ALSA: Jack: handle jack embedded kcontrol creating within ctljack') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/ctljack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/ctljack.c b/sound/core/ctljack.c index 8f8d1033425c..9149a4aefa95 100644 --- a/sound/core/ctljack.c +++ b/sound/core/ctljack.c @@ -70,7 +70,7 @@ snd_kctl_jack_new(const char *name, struct snd_card *card) return NULL; jack_kctl_name_gen(kctl->id.name, name, sizeof(kctl->id.name)); - kctl->id.index = get_available_index(card, name); + kctl->id.index = get_available_index(card, kctl->id.name); kctl->private_value = 0; return kctl; } -- 2.4.2

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Add mic volume fix quirk for Logitech Quickcam Fusion
by Wolfram Sang 29 May '15

29 May '15

Fix this from the logs: usb 7-1: New USB device found, idVendor=046d, idProduct=08ca ... usb 7-1: Warning! Unlikely big volume range (=3072), cval->res is probably wrong. usb 7-1: [5] FU [Mic Capture Volume] ch = 1, val = 4608/7680/1 Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> --- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 3e2ef61c627b83..4256fdedbd8009 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -918,6 +918,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ + case USB_ID(0x046d, 0x08ca): /* Logitech Quickcam Fusion */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. -- 2.1.4

2 1

Re: [alsa-devel] Add dell new platform support
by Takashi Iwai 29 May '15

29 May '15

At Fri, 29 May 2015 07:46:36 +0000, Kailang wrote: > > Hi Takashi, > > Dell new platform with ALC256 codec. SSID is 0x0706. > Attach patch is add the verb table quirk to enable headset mode. Thanks, applied. Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - Set patch_ops before calling auto-parser
by Takashi Iwai 29 May '15

29 May '15

... so that the parser code can overwrite some optional ops. For Realtek and others, it can be set solely in the spec allocator, so it results in a good code cleanup, too. With this change, we can finally remove the local stream_pm setup and rely on the generic parser's automatic setting. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 3 +-- sound/pci/hda/patch_ca0110.c | 3 +-- sound/pci/hda/patch_ca0132.c | 8 +++--- sound/pci/hda/patch_cirrus.c | 12 ++++----- sound/pci/hda/patch_cmedia.c | 4 +-- sound/pci/hda/patch_conexant.c | 3 +-- sound/pci/hda/patch_realtek.c | 61 ++++++++++++++++-------------------------- sound/pci/hda/patch_sigmatel.c | 20 +------------- sound/pci/hda/patch_via.c | 21 +++++---------- 9 files changed, 44 insertions(+), 91 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 231f89029779..c033a4ee6547 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -205,8 +205,6 @@ static int ad198x_parse_auto_config(struct hda_codec *codec, bool indep_hp) if (err < 0) return err; - codec->patch_ops = ad198x_auto_patch_ops; - return 0; } @@ -223,6 +221,7 @@ static int alloc_ad_spec(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; snd_hda_gen_spec_init(&spec->gen); + codec->patch_ops = ad198x_auto_patch_ops; return 0; } diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 447302695195..484bbf4134cd 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -63,6 +63,7 @@ static int patch_ca0110(struct hda_codec *codec) return -ENOMEM; snd_hda_gen_spec_init(spec); codec->spec = spec; + codec->patch_ops = ca0110_patch_ops; spec->multi_cap_vol = 1; codec->bus->needs_damn_long_delay = 1; @@ -71,8 +72,6 @@ static int patch_ca0110(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = ca0110_patch_ops; - return 0; error: diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 8dec790dae45..0f039abe9673 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4709,6 +4709,10 @@ static int patch_ca0132(struct hda_codec *codec) codec->spec = spec; spec->codec = codec; + codec->patch_ops = ca0132_patch_ops; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; + /* Detect codec quirk */ quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); if (quirk) @@ -4737,10 +4741,6 @@ static int patch_ca0132(struct hda_codec *codec) if (err < 0) return err; - codec->patch_ops = ca0132_patch_ops; - codec->pcm_format_first = 1; - codec->no_sticky_stream = 1; - return 0; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 50e9dd675579..25ccf781fbe7 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -584,6 +584,7 @@ static int patch_cs420x(struct hda_codec *codec) if (!spec) return -ENOMEM; + codec->patch_ops = cs_patch_ops; spec->gen.automute_hook = cs_automute; codec->single_adc_amp = 1; @@ -595,8 +596,6 @@ static int patch_cs420x(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = cs_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -738,6 +737,7 @@ static int patch_cs4208(struct hda_codec *codec) if (!spec) return -ENOMEM; + codec->patch_ops = cs_patch_ops; spec->gen.automute_hook = cs_automute; /* exclude NID 0x10 (HP) from output volumes due to different steps */ spec->gen.out_vol_mask = 1ULL << 0x10; @@ -756,8 +756,6 @@ static int patch_cs4208(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = cs_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -1150,6 +1148,7 @@ static int patch_cs4210(struct hda_codec *codec) if (!spec) return -ENOMEM; + codec->patch_ops = cs421x_patch_ops; spec->gen.automute_hook = cs_automute; snd_hda_pick_fixup(codec, cs421x_models, cs421x_fixup_tbl, @@ -1167,8 +1166,6 @@ static int patch_cs4210(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = cs421x_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -1187,11 +1184,12 @@ static int patch_cs4213(struct hda_codec *codec) if (!spec) return -ENOMEM; + codec->patch_ops = cs421x_patch_ops; + err = cs421x_parse_auto_config(codec); if (err < 0) goto error; - codec->patch_ops = cs421x_patch_ops; return 0; error: diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 617d9012e78a..f5ed078710f8 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -57,6 +57,7 @@ static int patch_cmi9880(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->patch_ops = cmi_auto_patch_ops; cfg = &spec->gen.autocfg; snd_hda_gen_spec_init(&spec->gen); @@ -67,7 +68,6 @@ static int patch_cmi9880(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = cmi_auto_patch_ops; return 0; error: @@ -86,6 +86,7 @@ static int patch_cmi8888(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->patch_ops = cmi_auto_patch_ops; cfg = &spec->gen.autocfg; snd_hda_gen_spec_init(&spec->gen); @@ -112,7 +113,6 @@ static int patch_cmi8888(struct hda_codec *codec) } } - codec->patch_ops = cmi_auto_patch_ops; return 0; error: diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 78b719b5b34d..f788a91b544a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -850,6 +850,7 @@ static int patch_conexant_auto(struct hda_codec *codec) return -ENOMEM; snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; + codec->patch_ops = cx_auto_patch_ops; cx_auto_parse_beep(codec); cx_auto_parse_eapd(codec); @@ -908,8 +909,6 @@ static int patch_conexant_auto(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = cx_auto_patch_ops; - /* Some laptops with Conexant chips show stalls in S3 resume, * which falls into the single-cmd mode. * Better to make reset, then. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index da2d37158d70..376444a9f11e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1003,6 +1003,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) codec->single_adc_amp = 1; /* FIXME: do we need this for all Realtek codec models? */ codec->spdif_status_reset = 1; + codec->patch_ops = alc_patch_ops; err = alc_codec_rename_from_preset(codec); if (err < 0) { @@ -1447,6 +1448,8 @@ static int patch_alc880(struct hda_codec *codec) spec->gen.need_dac_fix = 1; spec->gen.beep_nid = 0x01; + codec->patch_ops.unsol_event = alc880_unsol_event; + snd_hda_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl, alc880_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -1459,10 +1462,6 @@ static int patch_alc880(struct hda_codec *codec) if (!spec->gen.no_analog) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - codec->patch_ops.unsol_event = alc880_unsol_event; - - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -1699,6 +1698,8 @@ static int patch_alc260(struct hda_codec *codec) spec->gen.prefer_hp_amp = 1; spec->gen.beep_nid = 0x01; + spec->shutup = alc_eapd_shutup; + snd_hda_pick_fixup(codec, alc260_fixup_models, alc260_fixup_tbl, alc260_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -1711,9 +1712,6 @@ static int patch_alc260(struct hda_codec *codec) if (!spec->gen.no_analog) set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - spec->shutup = alc_eapd_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -2298,8 +2296,6 @@ static int patch_alc882(struct hda_codec *codec) if (!spec->gen.no_analog && spec->gen.beep_nid) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -2435,6 +2431,8 @@ static int patch_alc262(struct hda_codec *codec) spec = codec->spec; spec->gen.shared_mic_vref_pin = 0x18; + spec->shutup = alc_eapd_shutup; + #if 0 /* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is * under-run @@ -2460,9 +2458,6 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->gen.no_analog && spec->gen.beep_nid) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - spec->shutup = alc_eapd_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -2566,6 +2561,8 @@ static int patch_alc268(struct hda_codec *codec) spec = codec->spec; spec->gen.beep_nid = 0x01; + spec->shutup = alc_eapd_shutup; + snd_hda_pick_fixup(codec, alc268_fixup_models, alc268_fixup_tbl, alc268_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -2587,9 +2584,6 @@ static int patch_alc268(struct hda_codec *codec) (0 << AC_AMPCAP_MUTE_SHIFT)); } - codec->patch_ops = alc_patch_ops; - spec->shutup = alc_eapd_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -5672,6 +5666,12 @@ static int patch_alc269(struct hda_codec *codec) if (codec->core.vendor_id != 0x10ec0292) codec->power_save_node = 1; +#ifdef CONFIG_PM + codec->patch_ops.suspend = alc269_suspend; + codec->patch_ops.resume = alc269_resume; +#endif + spec->shutup = alc269_shutup; + snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups); @@ -5768,15 +5768,6 @@ static int patch_alc269(struct hda_codec *codec) if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid) set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - codec->patch_ops.stream_pm = snd_hda_gen_stream_pm; -#ifdef CONFIG_PM - codec->patch_ops.suspend = alc269_suspend; - codec->patch_ops.resume = alc269_resume; -#endif - if (!spec->shutup) - spec->shutup = alc269_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -5892,6 +5883,10 @@ static int patch_alc861(struct hda_codec *codec) spec = codec->spec; spec->gen.beep_nid = 0x23; +#ifdef CONFIG_PM + spec->power_hook = alc_power_eapd; +#endif + snd_hda_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -5903,11 +5898,6 @@ static int patch_alc861(struct hda_codec *codec) if (!spec->gen.no_analog) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); - codec->patch_ops = alc_patch_ops; -#ifdef CONFIG_PM - spec->power_hook = alc_power_eapd; -#endif - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -5984,6 +5974,8 @@ static int patch_alc861vd(struct hda_codec *codec) spec = codec->spec; spec->gen.beep_nid = 0x23; + spec->shutup = alc_eapd_shutup; + snd_hda_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -5995,10 +5987,6 @@ static int patch_alc861vd(struct hda_codec *codec) if (!spec->gen.no_analog) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - - spec->shutup = alc_eapd_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -6599,6 +6587,8 @@ static int patch_alc662(struct hda_codec *codec) spec = codec->spec; + spec->shutup = alc_eapd_shutup; + /* handle multiple HPs as is */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; @@ -6650,9 +6640,6 @@ static int patch_alc662(struct hda_codec *codec) } } - codec->patch_ops = alc_patch_ops; - spec->shutup = alc_eapd_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -6689,8 +6676,6 @@ static int patch_alc680(struct hda_codec *codec) return err; } - codec->patch_ops = alc_patch_ops; - return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6ff92aba0f57..bdcda6a41032 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4417,6 +4417,7 @@ static int alloc_stac_spec(struct hda_codec *codec) codec->spec = spec; codec->no_trigger_sense = 1; /* seems common with STAC/IDT codecs */ spec->gen.dac_min_mute = true; + codec->patch_ops = stac_patch_ops; return 0; } @@ -4433,7 +4434,6 @@ static int patch_stac9200(struct hda_codec *codec) spec->linear_tone_beep = 1; spec->gen.own_eapd_ctl = 1; - codec->patch_ops = stac_patch_ops; codec->power_filter = snd_hda_codec_eapd_power_filter; snd_hda_add_verbs(codec, stac9200_eapd_init); @@ -4466,8 +4466,6 @@ static int patch_stac925x(struct hda_codec *codec) spec->linear_tone_beep = 1; spec->gen.own_eapd_ctl = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_add_verbs(codec, stac925x_core_init); snd_hda_pick_fixup(codec, stac925x_models, stac925x_fixup_tbl, @@ -4537,8 +4535,6 @@ static int patch_stac92hd73xx(struct hda_codec *codec) spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_pick_fixup(codec, stac92hd73xx_models, stac92hd73xx_fixup_tbl, stac92hd73xx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -4614,8 +4610,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->default_polarity = -1; /* no default cfg */ - codec->patch_ops = stac_patch_ops; - snd_hda_add_verbs(codec, stac92hd83xxx_core_init); snd_hda_pick_fixup(codec, stac92hd83xxx_models, stac92hd83xxx_fixup_tbl, @@ -4664,8 +4658,6 @@ static int patch_stac92hd95(struct hda_codec *codec) spec->num_pwrs = ARRAY_SIZE(stac92hd95_pwr_nids); spec->default_polarity = 0; - codec->patch_ops = stac_patch_ops; - snd_hda_pick_fixup(codec, stac92hd95_models, stac92hd95_fixup_tbl, stac92hd95_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -4704,8 +4696,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->gen.mixer_nid = 0x17; spec->have_spdif_mux = 1; - codec->patch_ops = stac_patch_ops; - /* GPIO0 = EAPD */ spec->gpio_mask = 0x01; spec->gpio_dir = 0x01; @@ -4784,8 +4774,6 @@ static int patch_stac922x(struct hda_codec *codec) spec->linear_tone_beep = 1; spec->gen.own_eapd_ctl = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_add_verbs(codec, stac922x_core_init); /* Fix Mux capture level; max to 2 */ @@ -4841,8 +4829,6 @@ static int patch_stac927x(struct hda_codec *codec) spec->aloopback_shift = 0; spec->eapd_switch = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_pick_fixup(codec, stac927x_models, stac927x_fixup_tbl, stac927x_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -4904,8 +4890,6 @@ static int patch_stac9205(struct hda_codec *codec) /* Turn on/off EAPD per HP plugging */ spec->eapd_switch = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_pick_fixup(codec, stac9205_models, stac9205_fixup_tbl, stac9205_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -4977,8 +4961,6 @@ static int patch_stac9872(struct hda_codec *codec) spec->linear_tone_beep = 1; spec->gen.own_eapd_ctl = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_add_verbs(codec, stac9872_core_init); snd_hda_pick_fixup(codec, stac9872_models, stac9872_fixup_tbl, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 31a95cca015d..2c6c6727ca93 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -117,6 +117,8 @@ static void via_playback_pcm_hook(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream, int action); +static const struct hda_codec_ops via_patch_ops; /* defined below */ + static struct via_spec *via_new_spec(struct hda_codec *codec) { struct via_spec *spec; @@ -137,6 +139,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; codec->power_save_node = 1; spec->gen.power_down_unused = 1; + codec->patch_ops = via_patch_ops; return spec; } @@ -472,7 +475,6 @@ static const struct hda_codec_ops via_patch_ops = { .init = via_init, .free = via_free, .unsol_event = snd_hda_jack_unsol_event, - .stream_pm = snd_hda_gen_stream_pm, #ifdef CONFIG_PM .suspend = via_suspend, .check_power_status = via_check_power_status, @@ -651,6 +653,9 @@ static int patch_vt1708(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + /* override some patch_ops */ + codec->patch_ops.build_controls = vt1708_build_controls; + codec->patch_ops.build_pcms = vt1708_build_pcms; spec->gen.mixer_nid = 0x17; /* set jackpoll_interval while parsing the codec */ @@ -679,10 +684,6 @@ static int patch_vt1708(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; - codec->patch_ops = via_patch_ops; - codec->patch_ops.build_controls = vt1708_build_controls; - codec->patch_ops.build_pcms = vt1708_build_pcms; - /* clear jackpoll_interval again; it's set dynamically */ codec->jackpoll_interval = 0; @@ -707,8 +708,6 @@ static int patch_vt1709(struct hda_codec *codec) return err; } - codec->patch_ops = via_patch_ops; - return 0; } @@ -735,7 +734,6 @@ static int patch_vt1708B(struct hda_codec *codec) return err; } - codec->patch_ops = via_patch_ops; return 0; } @@ -800,7 +798,6 @@ static int patch_vt1708S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } @@ -842,7 +839,6 @@ static int patch_vt1702(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } @@ -915,7 +911,6 @@ static int patch_vt1718S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } @@ -1015,7 +1010,6 @@ static int patch_vt1716S(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = vt1716s_dmic_mixer; spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; - codec->patch_ops = via_patch_ops; return 0; } @@ -1123,7 +1117,6 @@ static int patch_vt2002P(struct hda_codec *codec) else spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } @@ -1162,7 +1155,6 @@ static int patch_vt1812(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1812_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } @@ -1200,7 +1192,6 @@ static int patch_vt3476(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } -- 2.4.2

1 0

[alsa-devel] [PATCH] ASoC: DAPM: Add APIs to create individual DAPM controls.
by Liam Girdwood 29 May '15

29 May '15

The topology core needs to be able to create individual widget controls at runtime and driver init. Add a regular locked and unlocked API calls to facilitate this requirement. The unlocked call is used by the topology core during component driver probing where the card dapm_mutex is held by the ASoC core and the locked version at non component driver probe time. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-dapm.c | 34 ++++++++++++++++++++++++++-------- 1 file changed, 26 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index defe0f0..62406d8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -52,8 +52,8 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, const char *control, int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink)); -static struct snd_soc_dapm_widget * -snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, +struct snd_soc_dapm_widget * +snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget); /* dapm power sequences - make this per codec in the future */ @@ -342,7 +342,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->value = template.on_val; - data->widget = snd_soc_dapm_new_control(widget->dapm, + data->widget = + snd_soc_dapm_new_control_unlocked(widget->dapm, &template); if (!data->widget) { kfree(data); @@ -3053,8 +3054,25 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); -static struct snd_soc_dapm_widget * +struct snd_soc_dapm_widget * snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget) +{ + struct snd_soc_dapm_widget *w; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + w = snd_soc_dapm_new_control_unlocked(dapm, widget); + if (!w) + dev_err(dapm->dev, + "ASoC: Failed to create DAPM control %s\n", + widget->name); + + mutex_unlock(&dapm->card->dapm_mutex); + return w; +} + +struct snd_soc_dapm_widget * +snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { struct snd_soc_dapm_widget *w; @@ -3199,7 +3217,7 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - w = snd_soc_dapm_new_control(dapm, widget); + w = snd_soc_dapm_new_control_unlocked(dapm, widget); if (!w) { dev_err(dapm->dev, "ASoC: Failed to create DAPM control %s\n", @@ -3437,7 +3455,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); - w = snd_soc_dapm_new_control(&card->dapm, &template); + w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); @@ -3488,7 +3506,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, dev_dbg(dai->dev, "ASoC: adding %s widget\n", template.name); - w = snd_soc_dapm_new_control(dapm, &template); + w = snd_soc_dapm_new_control_unlocked(dapm, &template); if (!w) { dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", dai->driver->playback.stream_name); @@ -3507,7 +3525,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, dev_dbg(dai->dev, "ASoC: adding %s widget\n", template.name); - w = snd_soc_dapm_new_control(dapm, &template); + w = snd_soc_dapm_new_control_unlocked(dapm, &template); if (!w) { dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", dai->driver->capture.stream_name); -- 2.1.4

2 1

[alsa-devel] [PATCH v3] ASoC: wm_adsp: Dump scratch registers on DSP shutdown
by Richard Fitzgerald 29 May '15

29 May '15

The SCRATCH registers are used by firmwares to hold diagnostic information. Log this during shutdown to assist analysis and debug of firmwares. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 27 +++++++++++++++++++++++++++ 1 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f6642c1..477390a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -121,6 +121,11 @@ #define ADSP2_WDMA_CONFIG_2 0x31 #define ADSP2_RDMA_CONFIG_1 0x34 +#define ADSP2_SCRATCH0 0x40 +#define ADSP2_SCRATCH1 0x41 +#define ADSP2_SCRATCH2 0x42 +#define ADSP2_SCRATCH3 0x43 + /* * ADSP2 Control */ @@ -364,6 +369,25 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, } } +static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) +{ + u16 scratch[4]; + int ret; + + ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0, + scratch, sizeof(scratch)); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); + return; + } + + adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", + be16_to_cpu(scratch[0]), + be16_to_cpu(scratch[1]), + be16_to_cpu(scratch[2]), + be16_to_cpu(scratch[3])); +} + static int wm_coeff_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1898,6 +1922,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + /* Log firmware state, it can be useful for analysis */ + wm_adsp2_show_fw_status(dsp); + dsp->running = false; regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, -- 1.7.2.5

2 1

Re: [alsa-devel] ASoC: PATCH: wm8731 - reinitialize regmap cache after hardware reset
by Mark Brown 29 May '15

29 May '15

On Thu, May 28, 2015 at 02:55:48PM -0700, Sergey Kiselev wrote: Please fix your mail client to word wrap within paragraphs, it makes your mail more readable. > On Thu, May 28, 2015 at 12:26 PM, Mark Brown wrote: > OK, I was trying to move the request supplies and reset code to > wm8731_{i2c|spi}_probe(). And I noticed the following weirdness with > the supplies: The current code in wm8731_probe() requests and enables > regulators, resets the codec hardware, tweaks some registers, and then > disables all the regulators: That's totally normal, the new register values will be rewritten as part of resyncing the register cache when the device is powered back up.

1 0

[alsa-devel] [PATCH] ALSA: fix building with CONFIG_SND_PROC_FS disabled
by Arnd Bergmann 29 May '15

29 May '15

A recent change to split the procfs support for ALSA turned out to be incomplete, leading to build errors: sound/core/seq/seq_info.c:61:12: error: redefinition of 'snd_seq_info_init' sound/core/seq/seq_device.c: In function 'seq_dev_proc_init': sound/core/seq/seq_device.c:276:2: error: 'info_entry' undeclared (first use in this function) sound/core/seq/seq_device.c: In function 'alsa_seq_device_exit': sound/core/seq/seq_device.c:309:22: error: 'info_entry' undeclared (first use in this function) This replaces some more instances of CONFIG_PROC_FS with CONFIG_SND_PROC_FS, as the original patch started. It is enough to avoid all randconfig regressions I saw on ARM because of the CONFIG_SND_PROC_FS introduction. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: cd6a65036f0e ("ALSA: replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS") --- I assume this was already reported. If not, please apply my patch or fold it into the original commit. diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index b29ffe835205..b65fa5a1943b 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -7,7 +7,7 @@ snd-seq-device-objs := seq_device.o snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \ seq_fifo.o seq_prioq.o seq_timer.o \ seq_system.o seq_ports.o -snd-seq-$(CONFIG_PROC_FS) += seq_info.o +snd-seq-$(CONFIG_SND_PROC_FS) += seq_info.o snd-seq-midi-objs := seq_midi.o snd-seq-midi-emul-objs := seq_midi_emul.o snd-seq-midi-event-objs := seq_midi_event.o diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 288af56a43cd..c4acf17e9f5e 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -272,7 +272,7 @@ EXPORT_SYMBOL_GPL(snd_seq_driver_unregister); static int __init seq_dev_proc_init(void) { -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_PROC_FS info_entry = snd_info_create_module_entry(THIS_MODULE, "drivers", snd_seq_root); if (info_entry == NULL) @@ -305,7 +305,7 @@ static void __exit alsa_seq_device_exit(void) #ifdef CONFIG_MODULES cancel_work_sync(&autoload_work); #endif -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_PROC_FS snd_info_free_entry(info_entry); #endif bus_unregister(&snd_seq_bus_type);

3 2

[alsa-devel] ASoC: codec register dump in debugfs
by Sergej Sawazki 29 May '15

29 May '15

Dear all, please correct me if I am wrong. Since the codec drivers have been converted to direct regmap API usage, it is not common to set the following snd_soc_codec_driver struct members: .reg_cache_size .reg_word_size However, the functions soc_codec_reg_show() and format_register_str() (responsible for formating and printing the codec register dumps to debugfs) depend on these members. I'm writing a ASoC machine driver. The board has two wm8741 codecs. I would like to see the codec register dumps in debugfs. eg: cat /sys/kernel/debug/asoc/<card_name>/codec\:wm8741.1-001a/codec_reg cat /sys/kernel/debug/asoc/<card_name>/codec\:wm8741.1-001b/codec_reg Is it OK to set the above mentioned members in a codec driver to "enable" the register dump output? Or should we refactor the above mentioned functions to use the regmap API? Or am I missing something? Thanks, Sergej

2 1

[alsa-devel] [BUG, bisect] ALSA: hda - No sound output
by Jeremiah Mahler 29 May '15

29 May '15

Takashi, all, On several of my machines there is no longer any sound output when running the latest linux-next (20150528). Activity can be seen in a mixer such as pavucontrol suggesting that there should be sound, but both the speakers and head phones are silent. I have bisected the kernel and found that the following patch introduced the bug. From 49fb18972581a781658a4637de76e6069ed5964e Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Wed, 27 May 2015 08:37:19 +0200 Subject: [PATCH] ALSA: hda - Set stream_pm ops automatically by generic parser This allows user to test power_save_node feature via sysfs or patch firmware even on the codecs that don't specify it. It'll also save a few lines. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 5 ++++- sound/pci/hda/patch_realtek.c | 1 - sound/pci/hda/patch_sigmatel.c | 1 - sound/pci/hda/patch_via.c | 1 - 4 files changed, 4 insertions(+), 4 deletions(-) -- - Jeremiah Mahler

2 2

[alsa-devel] [PATCH 00/10] Leftover CONFIG_PROC_FS fixes & cleanups
by Takashi Iwai 29 May '15

29 May '15

Hi, here is a patch series to fix / clean up the leftover CONFIG_PROC_FS after Keyon's patch to introduce CONFIG_SND_PROC_FS. Takashi === Takashi Iwai (10): ALSA: aloop: Drop unnecessary ifdef CONFIG_PROC_FS ALSA: dummy: Replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS ALSA: opl4: Fix / cleanup ifdef CONFIG_PROC_FS ALSA: ak4xxx-adda: Drop unnecessary ifdef CONFIG_PROC_FS ALSA: ac97: Fix ifdef CONFIG_PROC_FS ALSA: ca0106: Fix/cleanup ifdef CONFIG_PROC_FS ALSA: cs46xx: Fix old ifdef CONFIG_PROC_FS ALSA: emu10k1: Fix/cleanup ifdef CONFIG_PROC_FS ALSA: emux: Fix/cleanup old ifdef CONFIG_PROC_FS ALSA: pci: Drop superfluous ifdef CONFIG_PROC_FS include/sound/emux_synth.h | 2 +- sound/drivers/aloop.c | 8 -------- sound/drivers/dummy.c | 4 ++-- sound/drivers/opl4/Makefile | 3 ++- sound/drivers/opl4/opl4_lib.c | 4 ---- sound/drivers/opl4/opl4_local.h | 7 +++++-- sound/drivers/opl4/opl4_proc.c | 4 ---- sound/i2c/other/ak4xxx-adda.c | 4 ---- sound/pci/ac97/Makefile | 2 +- sound/pci/ac97/ac97_local.h | 2 +- sound/pci/ak4531_codec.c | 6 ------ sound/pci/atiixp.c | 4 ---- sound/pci/atiixp_modem.c | 4 ---- sound/pci/ca0106/Makefile | 3 ++- sound/pci/ca0106/ca0106_main.c | 2 +- sound/pci/ca0106/ca0106_proc.c | 4 ---- sound/pci/cmipci.c | 5 ----- sound/pci/cs46xx/cs46xx_lib.c | 4 ++-- sound/pci/cs46xx/cs46xx_lib.h | 4 ++-- sound/pci/cs46xx/dsp_spos.c | 4 ++-- sound/pci/cs46xx/dsp_spos_scb_lib.c | 6 +++--- sound/pci/emu10k1/Makefile | 3 ++- sound/pci/emu10k1/emu10k1_main.c | 2 +- sound/pci/emu10k1/emuproc.c | 2 -- sound/pci/ice1712/quartet.c | 7 +------ sound/pci/intel8x0.c | 4 ---- sound/pci/intel8x0m.c | 5 ----- sound/pci/oxygen/oxygen_lib.c | 4 ---- sound/synth/emux/Makefile | 5 +++-- sound/synth/emux/emux.c | 4 ---- sound/synth/emux/emux_proc.c | 4 ---- sound/synth/emux/emux_voice.h | 6 +++++- 32 files changed, 36 insertions(+), 96 deletions(-) -- 2.4.2

2 11

[alsa-devel] [PATCH] ALSA: core: Fix randconfig build wrt CONFIG_PROC_FS
by Takashi Iwai 29 May '15

29 May '15

There are a few leftover CONFIG_PROC_FS forgotten to replace with CONFIG_SND_PROC_FS. Fixes: cd6a65036f0e ('ALSA: replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS') Reported-by: Jim Davis <jim.epost(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/Makefile | 2 +- sound/core/seq/seq_device.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index b29ffe835205..b65fa5a1943b 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -7,7 +7,7 @@ snd-seq-device-objs := seq_device.o snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \ seq_fifo.o seq_prioq.o seq_timer.o \ seq_system.o seq_ports.o -snd-seq-$(CONFIG_PROC_FS) += seq_info.o +snd-seq-$(CONFIG_SND_PROC_FS) += seq_info.o snd-seq-midi-objs := seq_midi.o snd-seq-midi-emul-objs := seq_midi_emul.o snd-seq-midi-event-objs := seq_midi_event.o diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 288af56a43cd..c4acf17e9f5e 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -272,7 +272,7 @@ EXPORT_SYMBOL_GPL(snd_seq_driver_unregister); static int __init seq_dev_proc_init(void) { -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_PROC_FS info_entry = snd_info_create_module_entry(THIS_MODULE, "drivers", snd_seq_root); if (info_entry == NULL) @@ -305,7 +305,7 @@ static void __exit alsa_seq_device_exit(void) #ifdef CONFIG_MODULES cancel_work_sync(&autoload_work); #endif -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_PROC_FS snd_info_free_entry(info_entry); #endif bus_unregister(&snd_seq_bus_type); -- 2.4.2

1 0

[alsa-devel] [PATCH v2 1/3] ASoC: topology: Add topology UAPI header
by Liam Girdwood 29 May '15

29 May '15

The ASoC topology UAPI header defines the structures required to define any DSP firmware audio topology and control objects from userspace. The following objects are supported :- o kcontrols including TLV controls. o DAPM widgets and graph elements o Vendor bespoke objects. o Coefficient data o FE PCM capabilities and config. o BE link capabilities and config. o Codec <-> codec link capabilities and config. o Topology object manifest. The file format is simple and divided into blocks for each object type and each block has a header that defines it's size and type. Blocks can be in any order of type and can either all be in a single file or spread across more than one file. Blocks also have a group identifier ID so that they can be loaded and unloaded by ID. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dapm.h | 1 + include/uapi/sound/asoc.h | 388 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 389 insertions(+) create mode 100644 include/uapi/sound/asoc.h diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 0bc8364..4f1b417 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -15,6 +15,7 @@ #include <linux/types.h> #include <sound/control.h> +#include <sound/asoc.h> struct device; diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h new file mode 100644 index 0000000..1221520 --- /dev/null +++ b/include/uapi/sound/asoc.h @@ -0,0 +1,388 @@ +/* + * uapi/sound/asoc.h -- ALSA SoC Firmware Controls and DAPM + * + * Copyright (C) 2012 Texas Instruments Inc. + * Copyright (C) 2015 Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, + * algorithms, equalisers, DAIs, widgets etc. +*/ + +#ifndef __LINUX_UAPI_SND_ASOC_H +#define __LINUX_UAPI_SND_ASOC_H + +#include <linux/types.h> +#include <sound/asound.h> + +/* + * Maximum number of channels topology kcontrol can represent. + */ +#define SND_SOC_TPLG_MAX_CHAN 8 + +/* + * Maximum number of PCM formats capability + */ +#define SND_SOC_TPLG_MAX_FORMATS 16 + +/* + * Maximum number of PCM stream configs + */ +#define SND_SOC_TPLG_STREAM_CONFIG_MAX 8 + +/* individual kcontrol info types - can be mixed with other types */ +#define SND_SOC_TPLG_CTL_VOLSW 1 +#define SND_SOC_TPLG_CTL_VOLSW_SX 2 +#define SND_SOC_TPLG_CTL_VOLSW_XR_SX 3 +#define SND_SOC_TPLG_CTL_ENUM 4 +#define SND_SOC_TPLG_CTL_BYTES 5 +#define SND_SOC_TPLG_CTL_ENUM_VALUE 6 +#define SND_SOC_TPLG_CTL_RANGE 7 +#define SND_SOC_TPLG_CTL_STROBE 8 + + +/* individual widget kcontrol info types - can be mixed with other types */ +#define SND_SOC_TPLG_DAPM_CTL_VOLSW 64 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE 65 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT 66 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE 67 +#define SND_SOC_TPLG_DAPM_CTL_PIN 68 + +/* DAPM widget types - add new items to the end */ +#define SND_SOC_TPLG_DAPM_INPUT 0 +#define SND_SOC_TPLG_DAPM_OUTPUT 1 +#define SND_SOC_TPLG_DAPM_MUX 2 +#define SND_SOC_TPLG_DAPM_MIXER 3 +#define SND_SOC_TPLG_DAPM_PGA 4 +#define SND_SOC_TPLG_DAPM_OUT_DRV 5 +#define SND_SOC_TPLG_DAPM_ADC 6 +#define SND_SOC_TPLG_DAPM_DAC 7 +#define SND_SOC_TPLG_DAPM_SWITCH 8 +#define SND_SOC_TPLG_DAPM_PRE 9 +#define SND_SOC_TPLG_DAPM_POST 10 +#define SND_SOC_TPLG_DAPM_AIF_IN 11 +#define SND_SOC_TPLG_DAPM_AIF_OUT 12 +#define SND_SOC_TPLG_DAPM_DAI_IN 13 +#define SND_SOC_TPLG_DAPM_DAI_OUT 14 +#define SND_SOC_TPLG_DAPM_DAI_LINK 15 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK + +/* Header magic number and string sizes */ +#define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ + +/* string sizes */ +#define SND_SOC_TPLG_NUM_TEXTS 16 + +/* ABI version */ +#define SND_SOC_TPLG_ABI_VERSION 0x2 + +/* Max size of TLV data */ +#define SND_SOC_TPLG_TLV_SIZE 32 + +/* + * File and Block header data types. + * Add new generic and vendor types to end of list. + * Generic types are handled by the core whilst vendors types are passed + * to the component drivers for handling. + */ +#define SND_SOC_TPLG_TYPE_MIXER 1 +#define SND_SOC_TPLG_TYPE_BYTES 2 +#define SND_SOC_TPLG_TYPE_ENUM 3 +#define SND_SOC_TPLG_TYPE_DAPM_GRAPH 4 +#define SND_SOC_TPLG_TYPE_DAPM_WIDGET 5 +#define SND_SOC_TPLG_TYPE_DAI_LINK 6 +#define SND_SOC_TPLG_TYPE_PCM 7 +#define SND_SOC_TPLG_TYPE_MANIFEST 8 +#define SND_SOC_TPLG_TYPE_CODEC_LINK 9 +#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_CODEC_LINK + +/* vendor block IDs - please add new vendor types to end */ +#define SND_SOC_TPLG_TYPE_VENDOR_FW 1000 +#define SND_SOC_TPLG_TYPE_VENDOR_CONFIG 1001 +#define SND_SOC_TPLG_TYPE_VENDOR_COEFF 1002 +#define SND_SOC_TPLG_TYPEVENDOR_CODEC 1003 + +#define SND_SOC_TPLG_STREAM_PLAYBACK 0 +#define SND_SOC_TPLG_STREAM_CAPTURE 1 + +/* + * Block Header. + * This header preceeds all object and object arrays below. + */ +struct snd_soc_tplg_hdr { + __le32 magic; /* magic number */ + __le32 abi; /* ABI version */ + __le32 version; /* optional vendor specific version details */ + __le32 type; /* SND_SOC_TPLG_TYPE_ */ + __le32 size; /* size of this structure */ + __le32 vendor_type; /* optional vendor specific type info */ + __le32 payload_size; /* data bytes, excluding this header */ + __le32 index; /* identifier for block */ + __le32 count; /* number of elements in block */ +} __attribute__((packed)); + +/* + * Private data. + * All topology objects may have private data that can be used by the driver or + * firmware. Core will ignore this data. + */ +struct snd_soc_tplg_private { + __le32 size; /* in bytes of private data */ + char data[0]; +} __attribute__((packed)); + +/* + * Kcontrol TLV data. + */ +struct snd_soc_tplg_ctl_tlv { + __le32 size; /* in bytes aligned to 4 */ + __le32 numid; /* control element numeric identification */ + __le32 count; /* number of elem in data array */ + __le32 data[SND_SOC_TPLG_TLV_SIZE]; +} __attribute__((packed)); + +/* + * Kcontrol channel data + */ +struct snd_soc_tplg_channel { + __le32 size; /* in bytes of this structure */ + __le32 reg; + __le32 shift; + __le32 id; /* ID maps to Left, Right, LFE etc */ +} __attribute__((packed)); + +/* + * Kcontrol Operations IDs + */ +struct snd_soc_tplg_kcontrol_ops_id { + __le32 get; + __le32 put; + __le32 info; +} __attribute__((packed)); + +/* + * kcontrol header + */ +struct snd_soc_tplg_ctl_hdr { + __le32 size; /* in bytes of this structure */ + __le32 type; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 access; + struct snd_soc_tplg_kcontrol_ops_id ops; + __le32 tlv_size; /* non zero means control has TLV data */ +} __attribute__((packed)); + +/* + * Stream Capabilities + */ +struct snd_soc_tplg_stream_caps { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le64 formats[SND_SOC_TPLG_MAX_FORMATS]; /* supported formats SNDRV_PCM_FMTBIT_* */ + __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ + __le32 rate_min; /* min rate */ + __le32 rate_max; /* max rate */ + __le32 channels_min; /* min channels */ + __le32 channels_max; /* max channels */ + __le32 periods_min; /* min number of periods */ + __le32 periods_max; /* max number of periods */ + __le32 period_size_min; /* min period size bytes */ + __le32 period_size_max; /* max period size bytes */ + __le32 buffer_size_min; /* min buffer size bytes */ + __le32 buffer_size_max; /* max buffer size bytes */ +} __attribute__((packed)); + +/* + * FE or BE Stream configuration supported by SW/FW + */ +struct snd_soc_tplg_stream { + __le32 size; /* in bytes of this structure */ + __le64 format; /* SNDRV_PCM_FMTBIT_* */ + __le32 rate; /* SNDRV_PCM_RATE_* */ + __le32 period_bytes; /* size of period in bytes */ + __le32 buffer_bytes; /* size of buffer in bytes */ + __le32 channels; /* channels */ + __le32 tdm_slot; /* optional BE bitmask of supported TDM slots */ + __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ +} __attribute__((packed)); + +/* + * Duplex stream configuration supported by SW/FW. + */ +struct snd_soc_tplg_stream_config { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct snd_soc_tplg_stream playback; + struct snd_soc_tplg_stream capture; +} __attribute__((packed)); + +/* + * Manifest. List totals for each payload type. Not used in parsing, but will + * be passed to the component driver before any other objects in order for any + * global componnent resource allocations. + * + * File block representation for manifest :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_manifest | 1 | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_manifest { + __le32 size; /* in bytes of this structure */ + __le32 control_elems; /* number of control elements */ + __le32 widget_elems; /* number of widget elements */ + __le32 graph_elems; /* number of graph elements */ + __le32 dai_elems; /* number of DAI elements */ + __le32 dai_link_elems; /* number of DAI link elements */ +} __attribute__((packed)); + +/* + * Mixer kcontrol. + * + * File block representation for mixer kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_mixer_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_mixer_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 min; + __le32 max; + __le32 platform_max; + __le32 invert; + __le32 num_channels; + struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; + struct snd_soc_tplg_ctl_tlv tlv; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Enumerated kcontrol + * + * File block representation for enum kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_enum_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_enum_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 num_channels; + struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; + __le32 items; + __le32 mask; + __le32 count; + char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4]; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Bytes kcontrol + * + * File block representation for bytes kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_bytes_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_bytes_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 max; + __le32 mask; + __le32 base; + __le32 num_regs; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * DAPM Graph Element + * + * File block representation for DAPM graph elements :- + * +-------------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-------------------------------------+----+ + * | struct snd_soc_tplg_dapm_graph_elem | N | + * +-------------------------------------+----+ + */ +struct snd_soc_tplg_dapm_graph_elem { + char sink[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char control[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char source[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; +} __attribute__((packed)); + +/* + * DAPM Widget. + * + * File block representation for DAPM widget :- + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_dapm_widget | N | + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_enum_control | 0|1 | + * | struct snd_soc_tplg_mixer_control | 0|N | + * +-------------------------------------+-----+ + * + * Optional enum or mixer control can be appended to the end of each widget + * in the block. + */ +struct snd_soc_tplg_dapm_widget { + __le32 size; /* in bytes of this structure */ + __le32 id; /* SND_SOC_DAPM_CTL */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char sname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + __le32 reg; /* negative reg = no direct dapm */ + __le32 shift; /* bits to shift */ + __le32 mask; /* non-shifted mask */ + __u32 invert; /* invert the power bit */ + __u32 ignore_suspend; /* kept enabled over suspend */ + __u16 event_flags; + __u16 event_type; + __u16 num_kcontrols; + struct snd_soc_tplg_private priv; + /* + * kcontrols that relate to this widget + * follow here after widget private data + */ +} __attribute__((packed)); + +struct snd_soc_tplg_pcm_cfg_caps { + struct snd_soc_tplg_stream_caps caps; + struct snd_soc_tplg_stream_config configs[SND_SOC_TPLG_STREAM_CONFIG_MAX]; + __le32 num_configs; /* number of configs */ +} __attribute__((packed)); + +/* + * Describes SW/FW specific features of PCM or DAI link. + * + * File block representation for PCM/DAI-Link :- + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_dapm_pcm_dai | N | + * +-----------------------------------+-----+ + */ +struct snd_soc_tplg_pcm_dai { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 id; /* unique ID - used to match */ + __le32 playback; /* supports playback mode */ + __le32 capture; /* supports capture mode */ + __le32 compress; /* 1 = compressed; 0 = PCM */ + struct snd_soc_tplg_pcm_cfg_caps capconf[2]; /* capabilities and configs */ +} __attribute__((packed)); + +#endif -- 2.1.4

5 12

[alsa-devel] randconfig build error with next-20150528, in sound/core/seq/seq_info.c
by Jim Davis 28 May '15

28 May '15

Building with the attached random configuration file, sound/core/seq/seq_info.c:61:12: error: redefinition of ‘snd_seq_info_init’ int __init snd_seq_info_init(void) ^ In file included from sound/core/seq/seq_info.c:26:0: sound/core/seq/seq_info.h:36:19: note: previous definition of ‘snd_seq_info_init ’ was here static inline int snd_seq_info_init(void) { return 0; } ^ sound/core/seq/seq_info.c:77:12: error: redefinition of ‘snd_seq_info_done’ int __exit snd_seq_info_done(void) ^ In file included from sound/core/seq/seq_info.c:26:0: sound/core/seq/seq_info.h:37:19: note: previous definition of ‘snd_seq_info_done’ was here static inline int snd_seq_info_done(void) { return 0; }

1 0

[alsa-devel] [RFC 0/5] Add a gpio jack device
by Dylan Reid 28 May '15

28 May '15

The first three changes add a gpio audio jack device. This device can be used on systems that report headphone or mic plug through GPIOS. There can be 0-N of these devices created per board each can report one of several events. For example, this allows for a single jack for HP/Mic and a separate jack for line out. The last two patches are an example of the jack being used on Acer's Tegra Chromebook, those can be considered separately and might not be worth the churn. Dylan Reid (5): ALSA: Add jack types to dt-bindings ASoC: jack - make add_gpiods accept pre-filled descriptors ASoC: Add GPIO based jack device ASoC: tegra_max98090: Change nyan to use gpio-jack ARM: tegra: nyan: specify gpio-audio-jack device .../devicetree/bindings/sound/gpio-audio-jack.txt | 39 +++++ arch/arm/boot/dts/tegra124-nyan.dtsi | 22 ++- include/dt-bindings/sound/audio-jack-events.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/gpio-audio-jack.c | 191 +++++++++++++++++++++ sound/soc/soc-jack.c | 9 +- sound/soc/tegra/tegra_max98090.c | 107 +++--------- 8 files changed, 291 insertions(+), 92 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/gpio-audio-jack.txt create mode 100644 include/dt-bindings/sound/audio-jack-events.h create mode 100644 sound/soc/codecs/gpio-audio-jack.c -- 2.2.1.62.g3f15098

3 21

[alsa-devel] [PATCH] ASoC: wm_adsp: Dump DSP_SCRATCH1 on DSP shutdown
by Richard Fitzgerald 28 May '15

28 May '15

The DSP_SCRATCH1 register is used by firmwares to hold diagnostic information. Include this in the shutdown message - it can be useful for debugging. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 14 +++++++++++++- 1 files changed, 13 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f6642c1..9b7c03c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -121,6 +121,11 @@ #define ADSP2_WDMA_CONFIG_2 0x31 #define ADSP2_RDMA_CONFIG_1 0x34 +#define ADSP2_SCRATCH0 0x40 +#define ADSP2_SCRATCH1 0x41 +#define ADSP2_SCRATCH2 0x42 +#define ADSP2_SCRATCH3 0x43 + /* * ADSP2 Control */ @@ -1880,6 +1885,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; + unsigned int scratch1 = 0xFFFFFFFF; int ret; switch (event) { @@ -1898,6 +1904,12 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + /* Capture DSP_SCRATCH1, it can be useful for analysis */ + ret = regmap_read(dsp->regmap, dsp->base + ADSP2_SCRATCH1, + &scratch1); + if (ret) + adsp_err(dsp, "Failed to read SCRATCH1 %d\n", ret); + dsp->running = false; regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, @@ -1935,7 +1947,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, kfree(alg_region); } - adsp_dbg(dsp, "Shutdown complete\n"); + adsp_info(dsp, "Shutdown complete (SCRATCH1:0x%x)\n", scratch1); break; default: -- 1.7.2.5

2 5

[alsa-devel] [PATCH] ASoC: wm_adsp: Dump scratch registers on DSP shutdown
by Richard Fitzgerald 28 May '15

28 May '15

The SCRATCH registers are used by firmwares to hold diagnostic information. Log this during shutdown to assist analysis and debug of firmwares. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 27 +++++++++++++++++++++++++++ 1 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f6642c1..6e7a7000 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -121,6 +121,11 @@ #define ADSP2_WDMA_CONFIG_2 0x31 #define ADSP2_RDMA_CONFIG_1 0x34 +#define ADSP2_SCRATCH0 0x40 +#define ADSP2_SCRATCH1 0x41 +#define ADSP2_SCRATCH2 0x42 +#define ADSP2_SCRATCH3 0x43 + /* * ADSP2 Control */ @@ -364,6 +369,25 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, } } +static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) +{ + u16 scratch[4]; + int ret; + + ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0, + scratch, sizeof(scratch)); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); + return; + } + + adsp_info(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", + be16_to_cpu(scratch[0]), + be16_to_cpu(scratch[1]), + be16_to_cpu(scratch[2]), + be16_to_cpu(scratch[3])); +} + static int wm_coeff_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1898,6 +1922,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + /* Log firmware state, it can be useful for analysis */ + wm_adsp2_show_fw_status(dsp); + dsp->running = false; regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, -- 1.7.2.5

1 0

[alsa-devel] [PATCH] ASoC: Intel: remove unused function hsw_pcm_free_modules()
by Jie Yang 28 May '15

28 May '15

Remove the unused function hsw_pcm_free_modules() to fix the compling warning: sound/soc/intel/haswell/sst-haswell-pcm.c:923:13: warning: 'sw_pcm_free_modules' defined but not used [-Wunused-function] static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 15 --------------- 1 file changed, 15 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index bd96629..1557e37 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -920,21 +920,6 @@ err: return -ENODEV; } -static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) -{ - struct sst_hsw *hsw = pdata->hsw; - struct hsw_pcm_data *pcm_data; - int i; - - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; - sst_hsw_runtime_module_free(pcm_data->runtime); - } - if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) { - sst_hsw_runtime_module_free(pdata->runtime_waves); - } -} - static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm *pcm = rtd->pcm; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: arizona: Fix noise generator gain TLV
by Richard Fitzgerald 28 May '15

28 May '15

The Arizona codec drivers had an incorrect dB scaling for the noise generator gain that started at 0dB and went upwards. Actually the highest setting is 0dB. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 0c6d1bc..d476221 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -42,7 +42,7 @@ struct wm5102_priv { static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); -static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); static const struct wm_adsp_region wm5102_dsp1_regions[] = { diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fbaeddb..3ee6cfd 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -167,7 +167,7 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); -static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); #define WM5110_NG_SRC(name, base) \ diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index a4d1177..e7c81ba 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -40,7 +40,7 @@ struct wm8997_priv { static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); -static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); static const struct reg_default wm8997_sysclk_reva_patch[] = { -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Drop unused fields from struct hda_codec_preset
by Takashi Iwai 28 May '15

28 May '15

It's very unlikely that we'd need these fields out of sudden. Let's drop them. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_bind.c | 10 +--------- sound/pci/hda/hda_codec.h | 4 ---- 2 files changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 00aa31c5f08e..d5ac25cc7fee 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -27,15 +27,7 @@ static int hda_codec_match(struct hdac_device *dev, struct hdac_driver *drv) u32 id = codec->probe_id ? codec->probe_id : codec->core.vendor_id; for (preset = driver->preset; preset->id; preset++) { - u32 mask = preset->mask; - - if (preset->afg && preset->afg != codec->core.afg) - continue; - if (preset->mfg && preset->mfg != codec->core.mfg) - continue; - if (!mask) - mask = ~0; - if (preset->id == (id & mask) && + if (preset->id == id && (!preset->rev || preset->rev == codec->core.revision_id)) { codec->preset = preset; return 1; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index ed7e9cfd699a..6c572588f7e5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -82,11 +82,7 @@ struct hda_bus { */ struct hda_codec_preset { unsigned int id; - unsigned int mask; - unsigned int subs; - unsigned int subs_mask; unsigned int rev; - hda_nid_t afg, mfg; const char *name; int (*patch)(struct hda_codec *codec); }; -- 2.4.2

1 0

[alsa-devel] [PATCH 1/2] ASoC: atmel: simplify Kconfig
by Alexandre Belloni 28 May '15

28 May '15

Enclose the options in if SND_ATMEL_SOC ... endif to remove the dependency. Also remove the useless description for SND_ATMEL_SOC_SSC. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/Kconfig | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index e7d08806f3e9..93abe4e6d596 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -6,27 +6,22 @@ config SND_ATMEL_SOC the ATMEL SSC interface. You will also need to select the audio interfaces to support below. +if SND_ATMEL_SOC + config SND_ATMEL_SOC_PDC tristate - depends on SND_ATMEL_SOC config SND_ATMEL_SOC_DMA tristate - depends on SND_ATMEL_SOC select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC tristate - depends on SND_ATMEL_SOC - help - Say Y or M if you want to add support for codecs the - ATMEL SSC interface. You will also needs to select the individual - machine drivers to support below. config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" depends on ARCH_AT91 || COMPILE_TEST - depends on ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI + depends on ATMEL_SSC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -37,7 +32,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_ATMEL_SOC_WM8904 tristate "Atmel ASoC driver for boards using WM8904 codec" depends on ARCH_AT91 || COMPILE_TEST - depends on ATMEL_SSC && SND_ATMEL_SOC && I2C + depends on ATMEL_SSC && I2C select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8904 @@ -48,10 +43,11 @@ config SND_ATMEL_SOC_WM8904 config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" depends on ARCH_AT91 || COMPILE_TEST - depends on ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI + depends on ATMEL_SSC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8731 help Say Y if you want to add support for audio SoC on an at91sam9x5 based board that is using WM8731 codec. +endif -- 2.1.4

5 10

[alsa-devel] [PATCH] ALSA: hda - Disable Headphone Mic boost for ALC662
by David Henningsson 28 May '15

28 May '15

When headphone mic boost is above zero, some 10 - 20 second delay might occur before the headphone mic is operational. Therefore disable the headphone mic boost control (recording gain is sufficient even without it). (Note: this patch is not about the headset mic, it's about the less common mic-in only mode.) BugLink: https://bugs.launchpad.net/bugs/1454235 Suggested-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d0fa455..da2d371 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4240,6 +4240,11 @@ static void alc_fixup_headset_mode_alc662(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; spec->gen.hp_mic = 1; /* Mic-in is same pin as headphone */ + + /* Disable boost for mic-in permanently. (This code is only called + from quirks that guarantee that the headphone is at NID 0x1b.) */ + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000); + snd_hda_override_wcaps(codec, 0x1b, get_wcaps(codec, 0x1b) & ~AC_WCAP_IN_AMP); } else alc_fixup_headset_mode(codec, fix, action); } -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Disable Headphone Mic boost for ALC662
by David Henningsson 28 May '15

28 May '15

When headphone mic boost is above zero, some 10 - 20 second delay might occur before the headphone mic is operational. Therefore disable the headphone mic boost control (recording gain is sufficient even without it). (Note: this patch is not about the headset mic, it's about the less common mic-in only mode.) BugLink: https://bugs.launchpad.net/bugs/1454235 Suggested-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d0fa455..87605fc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4240,6 +4240,10 @@ static void alc_fixup_headset_mode_alc662(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; spec->gen.hp_mic = 1; /* Mic-in is same pin as headphone */ + + /* Disable boost for mic-in permanently */ + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000); + snd_hda_override_wcaps(codec, 0x1b, get_wcaps(codec, 0x1b) & ~AC_WCAP_IN_AMP); } else alc_fixup_headset_mode(codec, fix, action); } -- 1.9.1

2 3

[alsa-devel] Freescale iMX6SL SSI (I2S master mode) Rising edge vs Falling edge
by Xuebing Wang 28 May '15

28 May '15

Nicolin and alsa-devel community: Source code in fsl_ssi.c shows below in function fsl_ssi_set_dai_fmt(): ----------- switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: ... ... /* Data on rising edge of bclk, frame low, 1clk before data */ strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; break; ----------- According to iMX6SL reference manual, 'TSCKP = 1' means "Data clocked out on *falling* edge of bit clock." (for I2S master mode), rather than "Data on rising edge of bclk in the comments". This means this comment in the source code is *partially* WRONG, am I correct? However, RSCKP is = 1 (for receiving), it means "Data latched on rising edge of bit clock", which is correct. Also, I am not sure why there is inconsistency in iMX6SL reference manual (48.9 SSI Memory Map/Register Definition), that TSCKP = 1 (for transmitting) is FALLING_EDGE, but RSCKP = 1 (for receiving) is RISING_EDGE. Note: SSI (I2S master mode) works perfect on my hardware, I did not observe anything wrong. Thanks. -- Xuebing

2 2

[alsa-devel] [RFC v2 0/13] dw_hdmi cleanups, audio preparation, helpers and ahb audio support
by Russell King - ARM Linux 28 May '15

28 May '15

I'm sending this series for comments, and to allow people to update their code bases (for those who are using my AHB audio driver on iMX6). This is really several series, and I'm not expecting this to be ready for the upcoming merge window. That said, if anyone wants to say that they're happy with some of the initial six patches, I'm happy to queue those for David, if he's willing to take them. This series applies on top of my previous series, with the three patches from the rockchip guys applied, iow: drm: rockchip/dw_hdmi-rockchip: improve for HDMI electrical test drm: bridge/dw_hdmi: separate VLEVCTRL settting into platform driver drm: bridge/dw_hdmi: fixed codec style drm: bridge/dw_hdmi: adjust n/cts setting order drm: bridge/dw_hdmi: protect n/cts setting with a mutex drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() The first four patches are a few cleanups to the dw_hdmi driver. The next two patches introduce new interfaces to the dw_hdmi driver to support AHB audio, including the errata found on iMX6 which requires N to be programmed to zero when the audio FIFO is not full. I'm expecting some discussion with these last two as we try to work out how to deal with the two variants of audio support for this part (AHB audio vs I2S audio). The following three add various helpers to the DRM and ALSA subsystems for audio support which is non-specific to the AHB audio driver. For DRM, this is a helper macro to obtain the pointer into the short audio descriptors. For ALSA, it's a set of helpers to restrict an audio PCM device's capabilities according to the ELD, and to generate the IEC958 channel status data. If these helpers are acceptable, we can convert a number of drivers to them. The ELD helper isn't quite finished, but should be sufficient for initial testing. The last two patches add support for the dw_hdmi audio. This is an ALSA driver rather than an ASoC driver as it's a pretty simple affair. I couldn't find a suitable location in the sound/ subtree to place it, so I've placed it next to the bridge - yes, we probably need to find it a better home in the sound/ subtree, but it currently makes use of dw_hdmi.h in the directory it lives. drivers/gpu/drm/bridge/Kconfig | 11 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 566 +++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.h | 14 + drivers/gpu/drm/bridge/dw_hdmi.c | 321 +++++++++------- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + drivers/gpu/drm/imx/dw_hdmi-imx.c | 5 + include/drm/bridge/dw_hdmi.h | 7 + include/drm/drm_edid.h | 19 + include/sound/pcm_drm_eld.h | 6 + include/sound/pcm_iec958.h | 9 + sound/core/Kconfig | 6 + sound/core/Makefile | 2 + sound/core/pcm_drm_eld.c | 92 +++++ sound/core/pcm_iec958.c | 95 +++++ 15 files changed, 1029 insertions(+), 128 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

9 68

Re: [alsa-devel] [PATCH 1/3] ASoC: max98090: read micbias from device property
by Anish Kumar 28 May '15

28 May '15

> From: "Fang, Yang A" <yang.a.fang(a)intel.com> > > This patch reads max98090 micbias from acpi or dt > > Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> > --- > sound/soc/codecs/max98090.c | 9 ++++++++- > 1 file changed, 8 insertions(+), 1 deletion(-) > > diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c > index c230626..27df17e 100644 > --- a/sound/soc/codecs/max98090.c > +++ b/sound/soc/codecs/max98090.c > @@ -2419,6 +2419,8 @@ static int max98090_probe(struct snd_soc_codec *codec) > struct max98090_cdata *cdata; > enum max98090_type devtype; > int ret = 0; > + int err; > + unsigned int micbias; > > dev_dbg(codec->dev, "max98090_probe\n"); > > @@ -2503,8 +2505,13 @@ static int max98090_probe(struct snd_soc_codec *codec) > snd_soc_write(codec, M98090_REG_BIAS_CONTROL, > M98090_VCM_MODE_MASK); > > + err = device_property_read_u32(codec->dev, "max,micbias", &micbias); if (err || micbias < M98090_MBVSEL_2V2 || micbias > M98090_MBVSEL_2V8) micbias = M98090_MBVSEL_2V8; Just stick sanity as well.

2 1

[alsa-devel] [PATCH] ALSA: hda - Disable power_save_node for IDT92HD71bxx
by Takashi Iwai 27 May '15

27 May '15

We've got a regression report that 4.1-rc causes noises on a Dell laptop. Similar like Realtek codec, this seems also triggered by the recent power_save_node feature. As this kind of issue is quite hard to debug without actual hardware, disable the power_save_node flag for this codec as a workaround. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98971 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0de0fd95144a..6833c74ed6ff 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4696,7 +4696,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_save_node = 1; + /* disabled power_save_node since it causes noises on a Dell machine */ + /* codec->power_save_node = 1; */ spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; -- 2.4.1

1 0

[alsa-devel] ASoC: regmap_config.max_register value wrong?
by Sergej Sawazki 27 May '15

27 May '15

Dear all, The following snippet is from the wm8741 codec driver. Shouldn't the .max_register field be set to WM8741_REGISTER_COUNT-1 (10) instead of WM8741_MAX_REGISTER (0x20)? [...] static const struct regmap_config wm8741_regmap = { .reg_bits = 7, .val_bits = 9, .max_register = WM8741_MAX_REGISTER, [...] WM8741_MAX_REGISTER is the address of the last register. WM8741_REGISTER_COUNT is the number of registers. The regmap_config documentation says: "@max_register: Optional, specifies the maximum valid register index." So it should be the *register index* and not the register address. Do I understand it correctly? Many thanks in advance. Best regards, Sergej

3 3

[alsa-devel] [PATCH] ALSA: firewire-lib: fix buffer-over-run when detecting packet discontinuity
by Takashi Sakamoto 27 May '15

27 May '15

When detecting packet discontinuity, handle_in_packet() returns minus value and this value is assigned to unsigned int variable, then the variable has huge value. As a result, the variable causes buffer-over-run in handle_out_packet(). This brings invalid page request and system hangup. This commit fixes the bug to add a new argument into handle_in_packet() and the number of handled data blocks is assignd to it. The function return value is just used to check error. I also considered to change the type of local variable to 'int' in in_stream_callback(). This idea is based on type-conversion in C standard, while it may cause future problems when adding more works. Thus, I dropped this idea. Fixes: 6fc6b9ce41c6('ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets') Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 2b3e8b1..7bb988f 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -688,10 +688,10 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, } static int handle_in_packet(struct amdtp_stream *s, - unsigned int payload_quadlets, __be32 *buffer) + unsigned int payload_quadlets, __be32 *buffer, + unsigned int *data_blocks) { u32 cip_header[2]; - unsigned int data_blocks; unsigned int data_block_quadlets, data_block_counter, dbc_interval; struct snd_pcm_substream *pcm = NULL; bool lost; @@ -709,7 +709,7 @@ static int handle_in_packet(struct amdtp_stream *s, dev_info_ratelimited(&s->unit->device, "Invalid CIP header for AMDTP: %08X:%08X\n", cip_header[0], cip_header[1]); - data_blocks = 0; + *data_blocks = 0; goto end; } @@ -717,7 +717,7 @@ static int handle_in_packet(struct amdtp_stream *s, if (payload_quadlets < 3 || ((cip_header[1] & CIP_FDF_MASK) == (AMDTP_FDF_NO_DATA << CIP_FDF_SHIFT))) { - data_blocks = 0; + *data_blocks = 0; } else { data_block_quadlets = (cip_header[0] & CIP_DBS_MASK) >> CIP_DBS_SHIFT; @@ -731,12 +731,12 @@ static int handle_in_packet(struct amdtp_stream *s, if (s->flags & CIP_WRONG_DBS) data_block_quadlets = s->data_block_quadlets; - data_blocks = (payload_quadlets - 2) / data_block_quadlets; + *data_blocks = (payload_quadlets - 2) / data_block_quadlets; } /* Check data block counter continuity */ data_block_counter = cip_header[0] & CIP_DBC_MASK; - if (data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && + if (*data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && s->data_block_counter != UINT_MAX) data_block_counter = s->data_block_counter; @@ -746,10 +746,10 @@ static int handle_in_packet(struct amdtp_stream *s, } else if (!(s->flags & CIP_DBC_IS_END_EVENT)) { lost = data_block_counter != s->data_block_counter; } else { - if ((data_blocks > 0) && (s->tx_dbc_interval > 0)) + if ((*data_blocks > 0) && (s->tx_dbc_interval > 0)) dbc_interval = s->tx_dbc_interval; else - dbc_interval = data_blocks; + dbc_interval = *data_blocks; lost = data_block_counter != ((s->data_block_counter + dbc_interval) & 0xff); @@ -762,30 +762,30 @@ static int handle_in_packet(struct amdtp_stream *s, return -EIO; } - if (data_blocks > 0) { + if (*data_blocks > 0) { buffer += 2; pcm = ACCESS_ONCE(s->pcm); if (pcm) - s->transfer_samples(s, pcm, buffer, data_blocks); + s->transfer_samples(s, pcm, buffer, *data_blocks); if (s->midi_ports) - read_midi_messages(s, buffer, data_blocks); + read_midi_messages(s, buffer, *data_blocks); } if (s->flags & CIP_DBC_IS_END_EVENT) s->data_block_counter = data_block_counter; else s->data_block_counter = - (data_block_counter + data_blocks) & 0xff; + (data_block_counter + *data_blocks) & 0xff; end: if (queue_in_packet(s) < 0) return -EIO; if (pcm) - update_pcm_pointers(s, pcm, data_blocks); + update_pcm_pointers(s, pcm, *data_blocks); - return data_blocks; + return 0; } static void out_stream_callback(struct fw_iso_context *context, u32 cycle, @@ -853,8 +853,8 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle, break; } - data_blocks = handle_in_packet(s, payload_quadlets, buffer); - if (data_blocks < 0) { + if (handle_in_packet(s, payload_quadlets, buffer, + &data_blocks) < 0) { s->packet_index = -1; break; } -- 2.1.4

2 1

Re: [alsa-devel] [PATCH]: No sound when using software based mmap support and dmix plugin
by Takashi Iwai 27 May '15

27 May '15

At Wed, 13 May 2015 17:12:33 -0700, Yogesh Soni wrote: > > Hi all, > > I am trying to have dmix plugin work ( I succeeded) with the audio card > that does not have hardware mmap support and uses pcm_indirect method to > map software buffer for mmap feature. > > Most of the plugins were working quite well with mmap/pcm_indirect combo. > However, dmix plugin did not work (no sound). I debugged and realized that > appl_ptr in pcm_direct.h(snd_pcm_indirect_playback_transfer()) was always 0 > and hence nothing got copied to slave pcm buffer. I then discovered that > dmix operates in freewheeling-mode.. > > I have made a small workaround patch (I believe it can be cleaned up) in > case of free-wheeling mode, basically using sw_ready to hold bytes that > need to be copied. I used rec->sw_ready because it looked safest to use for > this purpose. As a result of this patch, normally period_size bytes are > copied to pcm buffer every period and sound is playing smooth. > > I am a noob in alsa and linux in general. So, I would like your opinion on > the patch. I would much prefer if I could write a patch that is generic and > clean enough, so that it could be used by others. patch is attached. Thanks for reporting and the patch. So, what actually prevents working with the current code in the free wheeling mode? Takashi

2 4

[alsa-devel] [PATCH] ALSA: hda - Fix noise on AMD radeon 290x controller
by Takashi Iwai 27 May '15

27 May '15

A new AMD controller [1002:aac8] seems to need the quirk for other AMD NS HDMI stuff, otherwise it gives noisy sounds. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99021 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 34040d26c94f..fea198c58196 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2089,6 +2089,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaab0), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaac8), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, -- 2.4.1

1 0

[alsa-devel] [PATCH RESEND v3 0/2] reducing memory footprint of ALSA
by Jie Yang 27 May '15

27 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... We plan to enable those reducing items only for EXPERT, and add knobs in sound/core/Kconfig. These first two patches is a start, adding SND_PROC_FS to config disable/enable ALSA proc FS, this disabling can save about 9KB code size on x86_64 platform. Changes in v3: 1. drop SND_REDUCED_MEMORY_FOOTPRINT and make configuration available only for EXPERT; 2. move the configure into sound/core/Kconfig; 3. Add missing Signed-off-by for PATCH 2/2. Changes in v2: 1. Prompt "Sound Proc FS Support" appears only when SND_REDUCED_MEMORY_FOOTPRINT is true. Jie Yang (2): ALSA: Kconfig: add config item SND_PROC_FS for expert ALSA: replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS include/sound/info.h | 8 ++++---- sound/core/Kconfig | 11 ++++++++++- sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 62 insertions(+), 53 deletions(-) -- 1.9.1

4 6

[alsa-devel] [PATCH] ASoC: codecs: use SNDRV_PCM_FMTBIT_* for format bitmask
by Maciej S. Szmigiero 27 May '15

27 May '15

snd_soc_pcm_stream.formats is a bitmask of SNDRV_PCM_FMTBIT_*, not of SNDRV_PCM_FORMAT_* (which are sequential integers), however some of ASoC CODEC drivers use these values instead. Found out by sparse on 0-day kernel tester. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index ee31fa7..38b3dad 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1186,16 +1186,16 @@ static struct snd_soc_dai_driver pm860x_dai[] = { .channels_min = 2, .channels_max = 2, .rates = PM860X_RATES, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .capture = { .stream_name = "PCM Capture", .channels_min = 2, .channels_max = 2, .rates = PM860X_RATES, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .ops = &pm860x_pcm_dai_ops, }, { @@ -1207,16 +1207,16 @@ static struct snd_soc_dai_driver pm860x_dai[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .capture = { .stream_name = "I2S Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .ops = &pm860x_i2s_dai_ops, }, diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 2341e8e..ed4cca7 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -320,7 +320,7 @@ static struct snd_soc_dai_driver stac9766_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, }, /* alsa ops */ .ops = &stac9766_dai_ops_digital, diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index e7d2ecd..493faf5 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -998,8 +998,8 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute) SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define WM8900_PCM_FORMATS \ - (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ - SNDRV_PCM_FORMAT_S24_LE) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm8900_dai_ops = { .hw_params = wm8900_hw_params, diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 9d18a0e..89cd2d6 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1054,8 +1054,8 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_48000) #define WM9713_PCM_FORMATS \ - (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ - SNDRV_PCM_FORMAT_S24_LE) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm9713_dai_ops_hifi = { .prepare = ac97_hifi_prepare,

4 7

[alsa-devel] [PATCH v3 0/2] reducing memory footprint of ALSA
by Jie Yang 27 May '15

27 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... We plan to enable those reducing items only for EXPERT, and add knobs in sound/core/Kconfig. These first two patches is a start, adding SND_PROC_FS to config disable/enable ALSA proc FS, this disabling can save about 9KB code size on x86_64 platform. Changes in v3: 1. drop SND_REDUCED_MEMORY_FOOTPRINT and make configuration available only for EXPERT; 2. move the configure into sound/core/Kconfig; 3. Add missing Signed-off-by for PATCH 2/2. Changes in v2: 1. Prompt "Sound Proc FS Support" appears only when SND_REDUCED_MEMORY_FOOTPRINT is true. Jie Yang (2): ALSA: Kconfig: add config item SND_PROC_FS for expert ALSA: replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS include/sound/info.h | 8 ++++---- sound/core/Kconfig | 9 +++++++++ sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 61 insertions(+), 52 deletions(-) -- 1.9.1

3 4

[alsa-devel] [PATCH 0/2] reducing memory footprint of ALSA
by Jie Yang 27 May '15

27 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... These first two patches is a start, add config menu in Kconfig, and disable ALSA proc FS once reducing memory footprint is selected, which can save about 9KB code size on x86_64 platform. Changes in v2: Prompt "Sound Proc FS Support" appears only when SND_REDUCED_MEMORY_FOOTPRINT is true. Jie Yang (2): ALSA: Kconfig: add config menu for reduced memory footprint ALSA: set no sound proc fs for reduced memory footprint include/sound/info.h | 8 ++++---- sound/Kconfig | 16 ++++++++++++++++ sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 68 insertions(+), 52 deletions(-) -- 1.9.1

4 10

[alsa-devel] [PATCH] ALSA: hda - Set stream_pm ops automatically by generic parser
by Takashi Iwai 27 May '15

27 May '15

This allows user to test power_save_node feature via sysfs or patch firmware even on the codecs that don't specify it. It'll also save a few lines. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 5 ++++- sound/pci/hda/patch_realtek.c | 1 - sound/pci/hda/patch_sigmatel.c | 1 - sound/pci/hda/patch_via.c | 1 - 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1c8678775f40..ac0db1679f09 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4926,9 +4926,12 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused || codec->power_save_node) + if (spec->power_down_unused || codec->power_save_node) { if (!codec->power_filter) codec->power_filter = snd_hda_gen_path_power_filter; + if (!codec->patch_ops.stream_pm) + codec->patch_ops.stream_pm = snd_hda_gen_stream_pm; + } if (!spec->no_analog && spec->beep_nid) { err = snd_hda_attach_beep_device(codec, spec->beep_nid); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fc0ccc78bdc7..e3bf72bb278c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5724,7 +5724,6 @@ static int patch_alc269(struct hda_codec *codec) set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); codec->patch_ops = alc_patch_ops; - codec->patch_ops.stream_pm = snd_hda_gen_stream_pm; #ifdef CONFIG_PM codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 43c99ce4a520..0de0fd95144a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4403,7 +4403,6 @@ static const struct hda_codec_ops stac_patch_ops = { #ifdef CONFIG_PM .suspend = stac_suspend, #endif - .stream_pm = snd_hda_gen_stream_pm, .reboot_notify = stac_shutup, }; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 31a95cca015d..742087ef378f 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -472,7 +472,6 @@ static const struct hda_codec_ops via_patch_ops = { .init = via_init, .free = via_free, .unsol_event = snd_hda_jack_unsol_event, - .stream_pm = snd_hda_gen_stream_pm, #ifdef CONFIG_PM .suspend = via_suspend, .check_power_status = via_check_power_status, -- 2.4.1

1 0

[alsa-devel] [PATCH] ASoC: fsl_spdif: Don't try to round-up for clock divisor calculation
by Nicolin Chen 27 May '15

27 May '15

As commit 6c8ca30eec7b ("ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation") mentioned that there's no more need to use a round up work around to get a better divisor since the clk-divider driver has been refined a lot. So this patch applies the same modification to fsl_spdif driver. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: Zidan Wang <zidan.wang(a)freescale.com> --- Fabio and Zidan, Theoretically, it should have the same problem as fsl_ssi driver had. But I don't have an S/PDIF test environment. So I need your helps to confirm it. Thank you. sound/soc/fsl/fsl_spdif.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 91eb3ae..8e93221 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -417,11 +417,9 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, if (clk != STC_TXCLK_SPDIF_ROOT) goto clk_set_bypass; - /* - * The S/PDIF block needs a clock of 64 * fs * txclk_df. - * So request 64 * fs * (txclk_df + 1) to get rounded. - */ - ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (txclk_df + 1)); + /* The S/PDIF block needs a clock of 64 * fs * txclk_df */ + ret = clk_set_rate(spdif_priv->txclk[rate], + 64 * sample_rate * txclk_df); if (ret) { dev_err(&pdev->dev, "failed to set tx clock rate\n"); return ret; @@ -1060,7 +1058,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * (txclk_df + 1) * 64; + rate_ideal = rate[index] * txclk_df * 64; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else -- 1.9.1

4 6

[alsa-devel] [PATCH v2] ASoC: Add GPIO based jack device
by Dylan Reid 27 May '15

27 May '15

Add a jack device that allows for separate headphone and mic detect GPIOs. This device will be used as an aux device and will registered the jack with the card at init time. The gpios are each optional allowing for a headphone, microphone, or combination devices to be created depending on the board configuration. A board will be able to have several of these jacks, one for each physical connection. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Tested on Acer Chromebook 13 Tegra device. I'm travelling this week, I'll try to get the rockchip team to integrate this with simple card for the 3288. Changes since v1: Static component_driver Coding style Improve error handling Get GPIOs in device probe to better handle EPROBE_DEFER --- .../devicetree/bindings/sound/gpio-audio-jack.txt | 39 ++++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/gpio-audio-jack.c | 201 +++++++++++++++++++++ 4 files changed, 247 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/gpio-audio-jack.txt create mode 100644 sound/soc/codecs/gpio-audio-jack.c diff --git a/Documentation/devicetree/bindings/sound/gpio-audio-jack.txt b/Documentation/devicetree/bindings/sound/gpio-audio-jack.txt new file mode 100644 index 0000000..5505a49 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gpio-audio-jack.txt @@ -0,0 +1,39 @@ +Gpio-Audio-Jack: + +GPIO based audio jacks. This can represent a headphone, mic, or combo jack. + +Required properties: + +- compatible : "linux,gpio-audio-jack" + +Optional properties: + +- gpio-audio-jack,jack-name : Name of the jack. Normally + 'Headphone Jack', 'Mic Jack', or + 'Headset Jack' +- gpio-audio-jack,sw-det-gpios : Reference to GPIOs that signals when + a jack is plugged. +- gpio-audio-jack,gpio-names : Names of the GPIOs. Must provide one + per sw-det-gpio entry. +- gpio-audio-jack,report-masks : Jack report mask from + dt-bindings/sound/audio-jack-events.h. + Must have one per sw-det-gpio entry. +- gpio-audio-jack,debounce-times : Debounce time in milliseconds for each + sw-det-gpio entry. + +Example of Headphone/Mic combo jack: + +audio_jack: gpio-audio-jack { + compatible = "linux,gpio-audio-jack"; + gpio-audio-jack,jack-name = "Headset Jack"; + gpio-audio-jack,sw-det-gpios = + <&gpio TEGRA_GPIO(I, 7) GPIO_ACTIVE_HIGH>, + <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_LOW>; + gpio-audio-jack,gpio-names = "Headphones", "Mic Jack"; + gpio-audio-jack,report-masks = <JACK_HEADPHONE>, <JACK_MICROPHONE>; + gpio-audio-jack,debounce-times = <150>, <150>; +}; + +sound { + nvidia,headset-dev = <&audio_jack>; + ... diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..4147593 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_GPIO_AUDIO_JACK select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,10 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_GPIO_AUDIO_JACK + tristate "GPIO based audio jack detection" + depends on GPIOLIB || COMPILE_TEST + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..ae570f5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-gpio-audio-jack-objs := gpio-audio-jack.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_GPIO_AUDIO_JACK) += snd-soc-gpio-audio-jack.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/gpio-audio-jack.c b/sound/soc/codecs/gpio-audio-jack.c new file mode 100644 index 0000000..29fdd8a --- /dev/null +++ b/sound/soc/codecs/gpio-audio-jack.c @@ -0,0 +1,201 @@ +/* + * GPIO based audio jack detection + * + * Copyright (C) 2015 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gpio.h> +#include <linux/init.h> +#include <linux/module.h> +#include <dt-bindings/sound/audio-jack-events.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/soc.h> + +#define JACK_NAME_CON_ID "gpio-audio-jack,jack-name" +#define GPIO_CON_ID "gpio-audio-jack,sw-det" +#define NAME_CON_ID "gpio-audio-jack,gpio-names" +#define REPORT_MASK_CON_ID "gpio-audio-jack,report-masks" +#define DEBOUNCE_CON_ID "gpio-audio-jack,debounce-times" + +struct gpio_audio_jack { + struct snd_soc_jack jack; + struct snd_soc_jack_gpio *gpios; + const char *jack_name; + int report_mask; + int gpio_count; +}; + +static int dt_jack_to_alsa_report(int dt_jack) +{ + switch (dt_jack) { + case JACK_HEADPHONE: + return SND_JACK_HEADPHONE; + case JACK_MICROPHONE: + return SND_JACK_MICROPHONE; + case JACK_LINEOUT: + return SND_JACK_LINEOUT; + case JACK_LINEIN: + return SND_JACK_LINEIN; + default: + return 0; + } +} + +static int gpio_audio_component_probe(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct gpio_audio_jack *priv = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_card_jack_new(component->card, priv->jack_name, + priv->report_mask, &priv->jack, NULL, 0); + if (ret) + return ret; + + return snd_soc_jack_add_gpiods(dev, &priv->jack, + priv->gpio_count, priv->gpios); +} + +static const struct snd_soc_component_driver gpio_jack_drv = { + .name = "gpio-audio-jack-dev", + .probe = gpio_audio_component_probe, +}; + +static int gpio_audio_jack_probe(struct platform_device *pdev) +{ + struct gpio_audio_jack *priv; + struct device *dev = &pdev->dev; + const char **gpio_names; + u32 *debounce; + u32 *gpio_report; + int ret; + int i; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + + ret = device_property_read_string(dev, JACK_NAME_CON_ID, + &priv->jack_name); + if (ret < 0) { + dev_err(dev, "Failed to read jack name %d\n", ret); + return ret; + } + + priv->gpio_count = gpiod_count(dev, GPIO_CON_ID); + if (priv->gpio_count <= 0) + return priv->gpio_count; + + gpio_names = devm_kcalloc(dev, priv->gpio_count, sizeof(*gpio_names), + GFP_KERNEL); + if (!gpio_names) + return -ENOMEM; + + gpio_report = devm_kcalloc(dev, priv->gpio_count, sizeof(*gpio_report), + GFP_KERNEL); + if (!gpio_report) + return -ENOMEM; + + debounce = devm_kcalloc(dev, priv->gpio_count, sizeof(*debounce), + GFP_KERNEL); + if (!debounce) + return -ENOMEM; + + priv->gpios = devm_kcalloc(dev, priv->gpio_count, sizeof(*priv->gpios), + GFP_KERNEL); + if (!priv->gpios) + return -ENOMEM; + + ret = device_property_read_string_array(dev, NAME_CON_ID, gpio_names, + priv->gpio_count); + if (ret < 0) { + dev_err(dev, "Failed to read gpio names %d\n", ret); + return ret; + } + + ret = device_property_read_u32_array(dev, REPORT_MASK_CON_ID, + gpio_report, priv->gpio_count); + if (ret < 0) { + dev_err(dev, "Failed to read gpio masks %d\n", ret); + return ret; + } + + ret = device_property_read_u32_array(dev, DEBOUNCE_CON_ID, + debounce, priv->gpio_count); + if (ret < 0) { + dev_err(dev, "Failed to read gpio debounce times %d\n", ret); + return ret; + } + + for (i = 0; i < priv->gpio_count; i++) { + priv->gpios[i].desc = gpiod_get_index(dev, GPIO_CON_ID, + i, GPIOD_IN); + if (IS_ERR(priv->gpios[i].desc)) { + ret = PTR_ERR(priv->gpios[i].desc); + dev_err(priv->gpios[i].gpiod_dev, + "Cannot get jack gpio at index %d: %d", + i, ret); + goto gpio_err; + } + + priv->gpios[i].name = gpio_names[i]; + priv->gpios[i].report = dt_jack_to_alsa_report(gpio_report[i]); + priv->gpios[i].debounce_time = debounce[i]; + priv->report_mask |= gpio_report[i]; + } + + devm_kfree(dev, gpio_names); + devm_kfree(dev, gpio_report); + devm_kfree(dev, debounce); + + return devm_snd_soc_register_component(&pdev->dev, &gpio_jack_drv, + NULL, 0); + +gpio_err: + for (i = 0; i < priv->gpio_count; i++) { + if (!IS_ERR(priv->gpios[i].desc)) + gpiod_put(priv->gpios[i].desc); + } + return ret; +} + +static int gpio_audio_jack_remove(struct platform_device *pdev) +{ + struct gpio_audio_jack *priv = dev_get_drvdata(&pdev->dev); + int i; + + for (i = 0; i < priv->gpio_count; i++) { + if (!IS_ERR(priv->gpios[i].desc)) + gpiod_put(priv->gpios[i].desc); + } + + return 0; +} + +static const struct of_device_id gpio_audio_of_match[] = { + { .compatible = "linux,gpio-audio-jack", }, + { } +}; +MODULE_DEVICE_TABLE(of, gpio_audio_of_match); + +static struct platform_driver gpio_audio_jack = { + .driver = { + .name = "gpio-audio-jack", + .of_match_table = gpio_audio_of_match, + }, + .probe = gpio_audio_jack_probe, + .remove = gpio_audio_jack_remove, +}; +module_platform_driver(gpio_audio_jack); + +MODULE_DESCRIPTION("ASoC GPIO audio jack driver"); +MODULE_AUTHOR("Dylan Reid <dgreid(a)chromium.org>"); +MODULE_LICENSE("GPL v2"); -- 2.2.1.62.g3f15098

2 1

[alsa-devel] [PATCH] ASoC: max98925: Fix mask for setting DAI invert mode
by Axel Lin 26 May '15

26 May '15

The M98925_DAI_WCI_MASK bit is not updated with current code. To properly set the DAI invert mode, the mask should be M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98925.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 9b5a17d..aad6642 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -346,7 +346,7 @@ static int max98925_dai_set_fmt(struct snd_soc_dai *codec_dai, } regmap_update_bits(max98925->regmap, MAX98925_FORMAT, - M98925_DAI_BCI_MASK, invert); + M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK, invert); return 0; } -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: Simplify format_register_str() without stack usages
by Takashi Iwai 26 May '15

26 May '15

Instead of allocating two string buffers on stack and copying them back, manipulate directly the target string buffer. This simplifies the code well. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-core.c | 27 +++++++++------------------ 1 file changed, 9 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 23732523f87c..ee86f84b7ccc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -92,30 +92,21 @@ static int format_register_str(struct snd_soc_codec *codec, int wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; int regsize = codec->driver->reg_word_size * 2; int ret; - char tmpbuf[len + 1]; - char regbuf[regsize + 1]; - - /* since tmpbuf is allocated on the stack, warn the callers if they - * try to abuse this function */ - WARN_ON(len > 63); /* +2 for ': ' and + 1 for '\n' */ if (wordsize + regsize + 2 + 1 != len) return -EINVAL; - ret = snd_soc_read(codec, reg); - if (ret < 0) { - memset(regbuf, 'X', regsize); - regbuf[regsize] = '\0'; - } else { - snprintf(regbuf, regsize + 1, "%.*x", regsize, ret); - } - - /* prepare the buffer */ - snprintf(tmpbuf, len + 1, "%.*x: %s\n", wordsize, reg, regbuf); - /* copy it back to the caller without the '\0' */ - memcpy(buf, tmpbuf, len); + sprintf(buf, "%.*x: ", wordsize, reg); + buf += wordsize + 2; + ret = snd_soc_read(codec, reg); + if (ret < 0) + memset(buf, 'X', regsize); + else + sprintf(buf, "%.*x", regsize, ret); + buf[regsize] = '\n'; + /* no NUL-termination needed */ return 0; } -- 2.4.1

2 1

Re: [alsa-devel] ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets
by Dan Carpenter 26 May '15

26 May '15

Hello Takashi Sakamoto, The patch 6fc6b9ce41c6: "ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets" from May 22, 2015, leads to the following static checker warning: sound/firewire/amdtp.c:857 in_stream_callback() warn: unsigned 'data_blocks' is never less than zero. sound/firewire/amdtp.c 855 856 data_blocks = handle_in_packet(s, payload_quadlets, buffer); 857 if (data_blocks < 0) { ^^^^^^^^^^^^^^^ Never true. But handle_in_packet() does return negative error codes. 858 s->packet_index = -1; 859 break; 860 } 861 regards, dan carpenter

2 1

[alsa-devel] [PATCH 0/2] reducing memory footprint of ALSA
by Jie Yang 26 May '15

26 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... These first two patches is a start, add config menu in Kconfig, and disable ALSA proc FS once reducing memory footprint is selected, which can save about 9KB code size on x86_64 platform. Comments are welcomed. ~Keyon Jie Yang (2): ALSA: Kconfig: add config menu for reduced memory footprint ALSA: set no sound proc fs for reduced memory footprint include/sound/info.h | 8 ++++---- sound/Kconfig | 16 ++++++++++++++++ sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 68 insertions(+), 52 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH 0/5] Trivial cleanups
by Takashi Iwai 26 May '15

26 May '15

Hi, here is a series of patches to clean up / fix the issues spotted by sparse. Takashi === Takashi Iwai (5): ALSA: dummy: make local data static ALSA: lx6464es: Fix duplicated SSID entries ALSA: lx6464es: Use NULL for pointers ALSA: bcd2000: Make local data static sound: sound_firmware: Fix invalid use of vfs_read() sound/drivers/dummy.c | 14 +++++++------- sound/pci/lx6464es/lx6464es.c | 16 ++++++++-------- sound/sound_firmware.c | 4 +--- sound/usb/bcd2000/bcd2000.c | 2 +- 4 files changed, 17 insertions(+), 19 deletions(-) -- 2.4.1

1 5

Re: [alsa-devel] ALC256 alias name for Dell
by Takashi Iwai 26 May '15

26 May '15

At Mon, 25 May 2015 09:25:07 +0000, Kailang wrote: > > Hi Takashi, > > Dell has new platform for ALC256 codec. > It calls ALC3246 for codec. Applied, thanks. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-alc256-alias-name-for-dell.patch <application/octet-stream (base64)>] >

1 0

Re: [alsa-devel] Bug 95261 - [SLK][Audio][HDA Analog] Audio capture is noisy if device is in RTD3 mode
by Takashi Iwai 26 May '15

26 May '15

At Mon, 25 May 2015 06:57:09 +0000, Pshou wrote: > > Hi Takashi, > This is when you shut up, occurred about five seconds record noise problems. > The problem, likely to occur in alc286 and sky board. I suppose the COEFs are specific to ALC286? If so, please either create alc286_shutup() or call COEF setups conditionally only for ALC286. thanks, Takashi > > Best regards > pshou > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 patch_realtek.diff <application/octet-stream (base64)>] >

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: topology
by Liam Girdwood 25 May '15

25 May '15

ASoC topology (formerly called dynamic firmware) allows audio DSP topology to be defined as part of the firmware package and not hard coded into the driver. This allows DSP vendors to ship multiple different firmwares or firmwares with programmable topologies for different devices (i.e. phones, tablets, TVs) using a single DSP driver. The DSP topology can be contained as part of the firmware file or exist as separate files. The topology core and file format is generic text and not tied to any single vendor or device. The topology core consists of kernel and userspace components :- Kernel. The kernel core reads in the topology data and builds the driver capabilities and audio topology based on this data. This includes building the DAPM graph, setting PCM and DAI capabilities, registering kcontrols and registering DAPM widgets. The userspace tool allows the creation of binary DSP FW topology files based on a topology text files. The tool is temporarily available on github and its intended to be merged into the alsa-utils package after a clean up. https://github.com/lgirdwood/topology.git Changes since RFC :- o Fixed patch series to survive bisection builds. o Removed widget ID enum from UAPI. o Added channel mappings to controls. o Added TLV macro patch to UAPI (should this be in asound.h ?). o Fixed locking for runtime and init time topology loading. o Moved the UAPI group index field to the block header. Changes since V1 (back in 2012) :- o Name changed to ASoC Topology o Structures from soc.h and soc-dapm.h no longer exposed as UAPIs o Moved to component model from codec/platform model. o Added reserved fields in each UAPI structure. o UAPI structures packed and endianess defined for each structure. o Added support for PCM, DAI and codec <-> codec link capabilities. o Added support for client drivers manifests. o Added support for private data for each dynamic object type. o Single list of dynamic objects rather than list of each object type. Liam

1 0

[alsa-devel] [PATCH 0/5 v2] ALSA: firewire-lib: purge restriction of synchronization for non-blocking mode
by Takashi Sakamoto 25 May '15

25 May '15

This patchset renews my previous one. [alsa-devel] [PATCH 0/4] firewire-lib: purge restriction of synchronization for non-blocking mode http://mailman.alsa-project.org/pipermail/alsa-devel/2015-May/092086.html Changes: * Use stack to keep calculation result per callback of isochronous receive context. * Fix my stupid mistake to compare byte count to quadlet count. (oh...) * Rename flag to proper name. * Add 4th patch newly for better reading. Takashi Sakamoto (5): ALSA: firewire-lib: add buffer-over-run protection at receiving more data blocks than expected ALSA: firewire-lib: simplify function to calculate the number of data blocks ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets ALSA: firewire-lib: set streaming error outside of packetization ALSA: firewire-lib: remove restriction for non-blocking mode sound/firewire/amdtp.c | 151 ++++++++++++++++++++++++-------------- sound/firewire/amdtp.h | 4 + sound/firewire/oxfw/oxfw-stream.c | 10 ++- 3 files changed, 106 insertions(+), 59 deletions(-) -- 2.1.4

2 7

[alsa-devel] [RESEND PATCH v1 0/2] Add dw_hdmi i2s audio driver support
by Yakir Yang 25 May '15

25 May '15

As Russell have send an series patches for dw_hdmi audio, it make us upload dw_hdmi audio patch more easier, and most important it works well on rockchip platform, so I make this patch that try to add another audio interface support in dw_hdmi driver. Here are the list that I have make in my series: - rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h - add dw_hdmi i2s audio driver Here are the series patches that I depends on: * drm: bridge/dw_hdmi: fixed codec style [Yakir Yang] * drm: bridge/dw_hdmi_rockchip: improve hdmi eye-diagram test * drm: bridge/dw_hdmi: improve hdmi single-end test * drm: imx/dw_hdmi: move phy comments [Russell King] * drm: bridge/dw_hdmi: clean up phy configuration * drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() * drm: bridge/dw_hdmi: protect n/cts setting with a mutex * drm: bridge/dw_hdmi: adjust n/cts setting order * drm: bridge/dw_hdmi: clean up hdmi_set_clk_regenerator() * drm: bridge/dw_hdmi: use drm_hdmi_avi_infoframe_from_display_mode() * drm: bridge/dw_hdmi: simplify hdmi_config_AVI() a little * drm: bridge/dw_hdmi: remove mhsyncpolarity/mvsyncpolarity/minterlaced * drm: bridge/dw_hdmi: introduce interface to setting sample rate * drm: bridge/dw_hdmi: introduce interfaces to enable and disable audio * drm/edid: add function to help find SADs * sound/core: add DRM ELD helper * sound/core: add IEC958 channel status helper * drm: bridge/dw_hdmi-ahb-audio: add audio driver * drm: bridge/dw_hdmi-ahb-audio: parse ELD from HDMI driver Best regards, Yakir Yang Yakir Yang (2): drm: bridge/dw_hdmi: rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h drm: bridge/dw_hdmi-i2s-audio: add audio driver drivers/gpu/drm/bridge/Kconfig | 9 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 2 +- .../{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} | 0 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c | 412 +++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi.c | 26 +- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + 7 files changed, 442 insertions(+), 11 deletions(-) rename drivers/gpu/drm/bridge/{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} (100%) create mode 100644 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c -- 2.1.2

3 4

[alsa-devel] ASoC: How to mix multi input ?
by Kuninori Morimoto 25 May '15

25 May '15

Hi Mark, ALSA SoC I would like to ask you how to mix multi input in ASoC. Renesas sound device(s) can mix multi input to 1 output. +-+ sound A --> |M| sound B --> |I| --> [ codec ] -> speaker sound C --> |X| +-+ Here, sound A/B/C will be used randomly. Does ALSA SoC already has this kind of system ? My image about these are like this. sound A: aplay -D hw:0 ... sound B: aplay -D hw:1 ... sound C: aplay -D hw:2 ... ... codec will be started if one of A/B/C are started, and it will be stopped if all of A/B/C are stopped.

4 8

[alsa-devel] [PATCH] ALSA: bebob: add Digidesign Mbox 2 Pro support
by Takashi Sakamoto 24 May '15

24 May '15

This device is based on DM1000E and BeBoB version 1 is installed. $ cat /proc/asound/cards 0 [Pro ]: BeBoB - Mbox 2 Pro DIGIDESIGN Mbox 2 Pro (id:1, rev:1), GUID 00a07e0100a90000 at fw1.0, S400 $ cat /proc/asound/Pro/firewire/firmware Manufacturer: bridgeCo Protocol Ver: 1 Build Ver: 0 GUID: 0x00A07E0100A90000 Model ID: 0x01 Model Rev: 1 Firmware Date: 20071031 Firmware Time: 034402 Firmware ID: 0xA9 Firmware Ver: 16777215 Base Addr: 0x20080000 Max Size: 1572864 Loader Date: 20051207 Loader Time: 205554 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 3 +++ 2 files changed, 4 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index ecec547..2a0a9aa 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -114,6 +114,7 @@ config SND_BEBOB * M-Audio FireWire410/AudioPhile/Solo * M-Audio Ozonic/NRV10/ProfireLightBridge * M-Audio FireWire 1814/ProjectMix IO + * Digidesign Mbox 2 Pro To compile this driver as a module, choose M here: the module will be called snd-bebob. diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 611b7da..e71bd35 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -57,6 +57,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define VEN_FOCUSRITE 0x0000130e #define VEN_MAUDIO1 0x00000d6c #define VEN_MAUDIO2 0x000007f5 +#define VEN_DIGIDESIGN 0x00a07e #define MODEL_FOCUSRITE_SAFFIRE_BOTH 0x00000000 #define MODEL_MAUDIO_AUDIOPHILE_BOTH 0x00010060 @@ -433,6 +434,8 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* M-Audio ProjectMix */ SND_BEBOB_DEV_ENTRY(VEN_MAUDIO1, MODEL_MAUDIO_PROJECTMIX, &maudio_special_spec), + /* Digidesign Mbox 2 Pro */ + SND_BEBOB_DEV_ENTRY(VEN_DIGIDESIGN, 0x0000a9, &spec_normal), /* IDs are unknown but able to be supported */ /* Apogee, Mini-ME Firewire */ /* Apogee, Mini-DAC Firewire */ -- 2.1.4

2 4

[alsa-devel] [PATCH 0/4] ALSA: firewire-lib: code cleanup
by Takashi Sakamoto 24 May '15

24 May '15

Current implementation includes several macros and functions with improper name, hard-coded numerical value and so on. This patchset is for code cleanup. Takashi Sakamoto (4): ALSA: firewire-lib: rename local functions for code cleanup ALSA: firewire-lib: macro arrangement for code cleanup ALSA: firewire-lib: use dev_err() when detecting incoming streaming error ALSA: firewire-lib: use protocol error when detecting wrong value in CIP header sound/firewire/amdtp.c | 104 ++++++++++++++++++++++++++----------------------- 1 file changed, 55 insertions(+), 49 deletions(-) -- 2.1.4

3 11

Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 24 May '15

24 May '15

Yet another experiment. Orpheus operation manual says there are six modes concerning Sync Source. -------------------------------------- Sync Source Master/Slave Description -------------------------------------- Local Master System is sync'ed to Orpheus' local clock Wordclock Master System is sync'ed to external wordclock applied to Orpheus DI Master System is sync'ed to external S/PDIF applied to Orpheus ADAT Master System is sync'ed to external ADAT (ADAT input must be enabled) PC DAW Slave Orpheus is slaved to system clock from host PC CSP Master System is sync'ed to Orpheus' 1394 clock (not recommended) -------------------------------------- With each of the five modes mentioned above (ADAT input not available), I tried to play music on Windows 7 using Audacity. The results are: ------------------- Local OK Wordclock NG DI NG PC DAW OK CSP OK ------------------- So it seems that if there is no external clock applied, when one is expected, Orpheus stops playing rather than trying to continue operation by free-running its clock. *** Takashi Sakamoto wrote: > Not funny, but headache. It's a customization by hardware vendor, I > mentioned. (in this case, by PrismSound Ltd.) > BridgeCo AG (currently ArchWave AG) produced DM1000/DM1100/DM1500 > chipset with BeBoB firmware. PrismSound Ltd. utilized the chipset and > firmware for streaming functionality of your Orpheus. > BeBoB firmware gives common ways to control itself, while the hadware > vendor added their own cusomization against it. I don't know exactly the > reason because there're few information about your device. I do not think I understand what you said here due to the lack of technical background, but what do you suggest at this moment to a casual applications programmer who wish to use snd-bebob with Orpheus? ------------ Answer 1 ------------ It should work. I received many successful reports for my final call for testing last year. It must be a hardware problem with your PC, or Firewire Cable, or Orpheus, or all of them. ------------ Answer 2 ------------ It should work. Check your software environment. Freshly installed Ubuntu 14.10 or later with the kernel 3.16 or later may not work. Always download the latest sources from GitHub, etc. ------------ Answer 3 ------------ Forget about snd-bebob, and fallback to libffado+jackd. The combination is loosing many packets, but if you say you do not notice any glitches, you should be completely happy with it. *** Regards, Mike Ood

2 2

[alsa-devel] [PATCH 1/3] ASoC: fsl_sai: add sai master mode support
by Zidan Wang 24 May '15

24 May '15

When sai works on master mode, set its bit clock and frame clock. SAI has 4 MCLK source, bus clock, MCLK1, MCLK2 and MCLK3. fsl_sai_set_bclk will select proper MCLK source, then calculate and set the bit clock divider. After fsl_sai_set_bclk, enable the selected mclk in hw_params(), and add hw_free() to disable the mclk. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 117 ++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 9 +++- 2 files changed, 121 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ee2671b..1ccc10d1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1,7 +1,7 @@ /* * Freescale ALSA SoC Digital Audio Interface (SAI) driver. * - * Copyright 2012-2013 Freescale Semiconductor, Inc. + * Copyright 2012-2015 Freescale Semiconductor, Inc. * * This program is free software, you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -251,12 +251,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 |= FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFM: + sai->is_slave_mode = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + sai->is_slave_mode = true; break; default: return -EINVAL; @@ -288,6 +290,79 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) return ret; } +static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + unsigned long clk_rate; + u32 savediv = 0, ratio, savesub = freq; + u32 id; + int ret = 0; + + /* Don't apply to slave mode */ + if (sai->is_slave_mode) + return 0; + + for (id = 0; id < FSL_SAI_MCLK_MAX; id++) { + clk_rate = clk_get_rate(sai->mclk_clk[id]); + if (!clk_rate) + continue; + + ratio = clk_rate / freq; + + ret = clk_rate - ratio * freq; + + /* + * Drop the source that can not be + * divided into the required rate. + */ + if (ret != 0 && clk_rate / ret < 1000) + continue; + + dev_dbg(dai->dev, + "ratio %d for freq %dHz based on clock %ldHz\n", + ratio, freq, clk_rate); + + if (ratio % 2 == 0 && ratio >= 2 && ratio <= 512) + ratio /= 2; + else + continue; + + if (ret < savesub) { + savediv = ratio; + sai->mclk_id[tx] = id; + savesub = ret; + } + + if (ret == 0) + break; + } + + if (savediv == 0) { + dev_err(dai->dev, "failed to derive required %cx rate: %d\n", + tx ? 'T' : 'R', freq); + return -EINVAL; + } + + if ((tx && sai->synchronous[TX]) || (!tx && !sai->synchronous[RX])) { + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } else { + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } + + dev_dbg(dai->dev, "best fit: clock id=%d, div=%d, deviation =%d\n", + sai->mclk_id[tx], savediv, savesub); + + return 0; +} + static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -297,6 +372,24 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(params); u32 word_width = snd_pcm_format_width(params_format(params)); u32 val_cr4 = 0, val_cr5 = 0; + int ret; + + if (!sai->is_slave_mode) { + ret = fsl_sai_set_bclk(cpu_dai, tx, + 2 * word_width * params_rate(params)); + if (ret) + return ret; + + /* Do not enable the clock if it is already enabled */ + if (!(sai->mclk_streams & BIT(substream->stream))) { + ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[tx]]); + if (ret) + return ret; + + sai->mclk_streams |= BIT(substream->stream); + } + + } if (!sai->is_dsp_mode) val_cr4 |= FSL_SAI_CR4_SYWD(word_width); @@ -322,6 +415,22 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int fsl_sai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + if (!sai->is_slave_mode && + sai->mclk_streams & BIT(substream->stream)) { + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]); + sai->mclk_streams &= ~BIT(substream->stream); + } + + return 0; +} + + static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -428,6 +537,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, .hw_params = fsl_sai_hw_params, + .hw_free = fsl_sai_hw_free, .trigger = fsl_sai_trigger, .startup = fsl_sai_startup, .shutdown = fsl_sai_shutdown, @@ -600,8 +710,9 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->bus_clk = NULL; } - for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { - sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[0] = sai->bus_clk; + for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i); sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 3466720..0662809 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -72,13 +72,15 @@ /* SAI Transmit and Recieve Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) -#define FSL_SAI_CR2_MSEL_MASK (0xff << 26) +#define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 #define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) #define FSL_SAI_CR2_MSEL_MCLK2 BIT(27) #define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27)) +#define FSL_SAI_CR2_MSEL(ID) ((ID) << 26) #define FSL_SAI_CR2_BCP BIT(25) #define FSL_SAI_CR2_BCD_MSTR BIT(24) +#define FSL_SAI_CR2_DIV_MASK 0xff /* SAI Transmit and Recieve Configuration 3 Register */ #define FSL_SAI_CR3_TRCE BIT(16) @@ -120,7 +122,7 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 -#define FSL_SAI_MCLK_MAX 3 +#define FSL_SAI_MCLK_MAX 4 /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 @@ -132,11 +134,14 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; + bool is_slave_mode; bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; + unsigned int mclk_id[2]; + unsigned int mclk_streams; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; -- 1.9.1

2 3

[alsa-devel] [PATCH trivial] ASoC: fsl_esai: Spelling s/specifially/specifically/
by Geert Uytterhoeven 24 May '15

24 May '15

Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 5c7597191e3ffc68..45eee131c47d43cd 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -517,7 +517,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, u32 bclk, mask, val; int ret; - /* Override slot_width if being specifially set */ + /* Override slot_width if being specifically set */ if (esai_priv->slot_width) slot_width = esai_priv->slot_width; -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for MS LifeCam HD-3000
by Vittorio G (VittGam) 24 May '15

24 May '15

Microsoft LifeCam HD-3000 (045e:0779) needs a similar quirk for suppressing the unsupported sample rate inquiry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481 Cc: <stable(a)vger.kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Vittorio Gambaletta <linuxbugs(a)vittgam.net> --- The dmesg is showing this every time the ALSA audio is opened by the application: [3759907.370000] 2:3:1: cannot get freq at ep 0x82 And in fact I had to initialize ALSA and v4l2 in a specific order in my application in order for them to both work. diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 46facfc..2917534 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1118,6 +1118,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema */ case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ + case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ return true; }

2 1

Re: [alsa-devel] [PATCH v2 1/3] ASoC: zx: Add ZTE zx296702 pcm support
by Jun Nie 23 May '15

23 May '15

2015-05-06 20:38 GMT+08:00 Jun Nie <jun.nie(a)linaro.org>: > This patch adds zx296702 ASoC DMA audio support. It > handle audio dma based on soc-dmaengine > Mark, Do you have any comments on this patch set? Or you suggest to add HDMI API calling in the same serial patch? For the HDMI calling, I have no good idea except direct call currently. Because I need dts hook to audio driver, while no HDMI call back data can be shared from video side to audio side in this independent initialization case. Jun > Signed-off-by: Jun Nie <jun.nie(a)linaro.org> > --- > sound/soc/Kconfig | 1 + > sound/soc/Makefile | 1 + > sound/soc/zte/Kconfig | 8 ++++++++ > sound/soc/zte/Makefile | 1 + > sound/soc/zte/zx296702-pcm.c | 46 ++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/zte/zx296702-pcm.h | 6 ++++++ > 6 files changed, 63 insertions(+) > create mode 100644 sound/soc/zte/Kconfig > create mode 100644 sound/soc/zte/Makefile > create mode 100644 sound/soc/zte/zx296702-pcm.c > create mode 100644 sound/soc/zte/zx296702-pcm.h > > diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig > index 3ba52da..e2828e1 100644 > --- a/sound/soc/Kconfig > +++ b/sound/soc/Kconfig > @@ -57,6 +57,7 @@ source "sound/soc/tegra/Kconfig" > source "sound/soc/txx9/Kconfig" > source "sound/soc/ux500/Kconfig" > source "sound/soc/xtensa/Kconfig" > +source "sound/soc/zte/Kconfig" > > # Supported codecs > source "sound/soc/codecs/Kconfig" > diff --git a/sound/soc/Makefile b/sound/soc/Makefile > index 974ba70..57bf32d 100644 > --- a/sound/soc/Makefile > +++ b/sound/soc/Makefile > @@ -38,3 +38,4 @@ obj-$(CONFIG_SND_SOC) += tegra/ > obj-$(CONFIG_SND_SOC) += txx9/ > obj-$(CONFIG_SND_SOC) += ux500/ > obj-$(CONFIG_SND_SOC) += xtensa/ > +obj-$(CONFIG_SND_SOC) += zte/ > diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig > new file mode 100644 > index 0000000..392ffa6 > --- /dev/null > +++ b/sound/soc/zte/Kconfig > @@ -0,0 +1,8 @@ > +config SND_SOC_ZX296702 > + tristate "SoC Audio for the ZX296702 System-on-Chip" > + depends on SOC_ZX296702 || COMPILE_TEST > + depends on COMMON_CLK > + select SND_SOC_GENERIC_DMAENGINE_PCM > + help > + Say Y or M here if you want support for SoC audio on ZX296702 > + chip. > diff --git a/sound/soc/zte/Makefile b/sound/soc/zte/Makefile > new file mode 100644 > index 0000000..f2a6af8 > --- /dev/null > +++ b/sound/soc/zte/Makefile > @@ -0,0 +1 @@ > +obj-$(CONFIG_SND_SOC_ZX296702) += zx296702-pcm.o > diff --git a/sound/soc/zte/zx296702-pcm.c b/sound/soc/zte/zx296702-pcm.c > new file mode 100644 > index 0000000..6e1a61a > --- /dev/null > +++ b/sound/soc/zte/zx296702-pcm.c > @@ -0,0 +1,46 @@ > +/* > + * Copyright (C) 2015 Linaro > + * > + * Author: Jun Nie <jun.nie(a)linaro.org> > + * > + * License terms: GNU General Public License (GPL) version 2 > + */ > + > +#include <linux/device.h> > +#include <linux/init.h> > +#include <linux/module.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/soc.h> > +#include <sound/dmaengine_pcm.h> > + > +#include "zx296702-pcm.h" > + > +static const struct snd_pcm_hardware snd_zx_hardware = { > + .info = SNDRV_PCM_INFO_MMAP | > + SNDRV_PCM_INFO_MMAP_VALID | > + SNDRV_PCM_INFO_INTERLEAVED | > + SNDRV_PCM_INFO_PAUSE | > + SNDRV_PCM_INFO_RESUME | > + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, > + .period_bytes_min = 32, > + .period_bytes_max = 16 * 1024, > + .periods_min = 2, > + .periods_max = 32, > + .buffer_bytes_max = 64 * 1024, > +}; > + > +static const struct snd_dmaengine_pcm_config zx_dmaengine_pcm_config = { > + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, > + .pcm_hardware = &snd_zx_hardware, > + .prealloc_buffer_size = 64 * 1024, > +}; > + > +int zx_pcm_platform_register(struct device *dev) > +{ > + return devm_snd_dmaengine_pcm_register(dev, &zx_dmaengine_pcm_config, > + 0); > +} > +EXPORT_SYMBOL_GPL(zx_pcm_platform_register); > + > +MODULE_LICENSE("GPL"); > diff --git a/sound/soc/zte/zx296702-pcm.h b/sound/soc/zte/zx296702-pcm.h > new file mode 100644 > index 0000000..4809a38 > --- /dev/null > +++ b/sound/soc/zte/zx296702-pcm.h > @@ -0,0 +1,6 @@ > +#ifndef _ZX_PCM_H > +#define _ZX_PCM_H > + > +int zx_pcm_platform_register(struct device *dev); > + > +#endif > -- > 1.9.1 >

2 4

Re: [alsa-devel] [PATCH v2 3/3] dt: Add documentation for the ZTE SPDIF controller
by Mark Brown 23 May '15

23 May '15

On Wed, May 06, 2015 at 08:38:05PM +0800, Jun Nie wrote: > This patch adds the devicetree documentation for the ZTE > zx296702 SPDIF audio controller. ...and here it is! Sorry - the DT binding is normally before the code in the patch series. This is fine, it's just patch 1 that needs to be resolved then.

2 1

[alsa-devel] [PATCH 1/2] ASoC: zx: Add ZTE zx296702 I2S DAI driver
by Jun Nie 23 May '15

23 May '15

Add ZTE zx296702 I2S DAI driver Signed-off-by: Jun Nie <jun.nie(a)linaro.org> --- sound/soc/zte/Kconfig | 7 + sound/soc/zte/Makefile | 1 + sound/soc/zte/zx296702-i2s.c | 437 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 445 insertions(+) create mode 100644 sound/soc/zte/zx296702-i2s.c diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig index a08f94c..b3fffe8 100644 --- a/sound/soc/zte/Kconfig +++ b/sound/soc/zte/Kconfig @@ -13,3 +13,10 @@ config ZX296702_SPDIF help Say Y or M if you want to add support for codecs attached to the zx296702 spdif interface + +config ZX296702_I2S + bool "ZX296702 i2s" + select ZX_PCM_DMA + help + Say Y or M if you want to add support for codecs attached to the + zx296702 i2s interface diff --git a/sound/soc/zte/Makefile b/sound/soc/zte/Makefile index 6eafade..6791aeb 100644 --- a/sound/soc/zte/Makefile +++ b/sound/soc/zte/Makefile @@ -1,2 +1,3 @@ obj-$(CONFIG_SND_SOC_ZX296702) += zx296702-pcm.o obj-$(CONFIG_ZX296702_SPDIF) += zx296702-spdif.o +obj-$(CONFIG_ZX296702_I2S) += zx296702-i2s.o diff --git a/sound/soc/zte/zx296702-i2s.c b/sound/soc/zte/zx296702-i2s.c new file mode 100644 index 0000000..0688256 --- /dev/null +++ b/sound/soc/zte/zx296702-i2s.c @@ -0,0 +1,437 @@ +/* + * Copyright (C) 2015 Linaro + * + * Author: Jun Nie <jun.nie(a)linaro.org> + * + * License terms: GNU General Public License (GPL) version 2 + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include <sound/core.h> +#include <sound/dmaengine_pcm.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "zx296702-pcm.h" + +#define ZX_I2S_PROCESS_CTRL 0x04 +#define ZX_I2S_TIMING_CTRL 0x08 +#define ZX_I2S_FIFO_CTRL 0x0C +#define ZX_I2S_FIFO_STATUS 0x10 +#define ZX_I2S_INT_EN 0x14 +#define ZX_I2S_INT_STATUS 0x18 +#define ZX_I2S_DATA 0x1C +#define ZX_I2S_FRAME_CNTR 0x20 + +#define I2S_DEAGULT_FIFO_THRES (0x10) +#define I2S_MAX_FIFO_THRES (0x20) + +#define ZX_I2S_PROCESS_TX_EN (1 << 0) +#define ZX_I2S_PROCESS_TX_DIS (0 << 0) +#define ZX_I2S_PROCESS_RX_EN (1 << 1) +#define ZX_I2S_PROCESS_RX_DIS (0 << 1) +#define ZX_I2S_PROCESS_I2S_EN (1 << 2) +#define ZX_I2S_PROCESS_I2S_DIS (0 << 2) + +#define ZX_I2S_TIMING_MAST (1 << 0) +#define ZX_I2S_TIMING_SLAVE (0 << 0) +#define ZX_I2S_TIMING_MS_MASK (1 << 0) +#define ZX_I2S_TIMING_LOOP (1 << 1) +#define ZX_I2S_TIMING_NOR (0 << 1) +#define ZX_I2S_TIMING_LOOP_MASK (1 << 1) +#define ZX_I2S_TIMING_PTNR (1 << 2) +#define ZX_I2S_TIMING_NTPR (0 << 2) +#define ZX_I2S_TIMING_PHASE_MASK (1 << 2) +#define ZX_I2S_TIMING_TDM (1 << 3) +#define ZX_I2S_TIMING_I2S (0 << 3) +#define ZX_I2S_TIMING_TIMING_MASK (1 << 3) +#define ZX_I2S_TIMING_LONG_SYNC (1 << 4) +#define ZX_I2S_TIMING_SHORT_SYNC (0 << 4) +#define ZX_I2S_TIMING_SYNC_MASK (1 << 4) +#define ZX_I2S_TIMING_TEAK_EN (1 << 5) +#define ZX_I2S_TIMING_TEAK_DIS (0 << 5) +#define ZX_I2S_TIMING_TEAK_MASK (1 << 5) +#define ZX_I2S_TIMING_STD_I2S (0 << 6) +#define ZX_I2S_TIMING_MSB_JUSTIF (1 << 6) +#define ZX_I2S_TIMING_LSB_JUSTIF (2 << 6) +#define ZX_I2S_TIMING_ALIGN_MASK (3 << 6) +#define ZX_I2S_TIMING_CHN_MASK (7 << 8) +#define ZX_I2S_TIMING_CHN(x) ((x - 1) << 8) +#define ZX_I2S_TIMING_LANE_MASK (3 << 11) +#define ZX_I2S_TIMING_LANE(x) ((x - 1) << 11) +#define ZX_I2S_TIMING_TSCFG_MASK (7 << 13) +#define ZX_I2S_TIMING_TSCFG(x) (x << 13) +#define ZX_I2S_TIMING_TS_WIDTH_MASK (0x1f << 16) +#define ZX_I2S_TIMING_TS_WIDTH(x) ((x - 1) << 16) +#define ZX_I2S_TIMING_DATA_SIZE_MASK (0x1f << 21) +#define ZX_I2S_TIMING_DATA_SIZE(x) ((x - 1) << 21) +#define ZX_I2S_TIMING_CFG_ERR_MASK (1 << 31) + +#define ZX_I2S_FIFO_CTRL_TX_RST (1 << 0) +#define ZX_I2S_FIFO_CTRL_TX_RST_MASK (1 << 0) +#define ZX_I2S_FIFO_CTRL_RX_RST (1 << 1) +#define ZX_I2S_FIFO_CTRL_RX_RST_MASK (1 << 1) +#define ZX_I2S_FIFO_CTRL_TX_DMA_EN (1 << 4) +#define ZX_I2S_FIFO_CTRL_TX_DMA_DIS (0 << 4) +#define ZX_I2S_FIFO_CTRL_TX_DMA_MASK (1 << 4) +#define ZX_I2S_FIFO_CTRL_RX_DMA_EN (1 << 5) +#define ZX_I2S_FIFO_CTRL_RX_DMA_DIS (0 << 5) +#define ZX_I2S_FIFO_CTRL_RX_DMA_MASK (1 << 5) +#define ZX_I2S_FIFO_CTRL_TX_THRES_MASK (0x1F << 8) +#define ZX_I2S_FIFO_CTRL_RX_THRES_MASK (0x1F << 16) + +#define CLK_RAT (32 * 4) + +struct zx_i2s_info { + struct snd_dmaengine_dai_dma_data dma_playback; + struct snd_dmaengine_dai_dma_data dma_capture; + struct clk *dai_clk; + void __iomem *reg_base; + int master; + resource_size_t mapbase; +}; + +static void zx_i2s_tx_en(void __iomem *base, bool on) +{ + unsigned long con; + + con = readl_relaxed(base + ZX_I2S_PROCESS_CTRL); + if (on) + con |= ZX_I2S_PROCESS_TX_EN | ZX_I2S_PROCESS_I2S_EN; + else + con &= ~(ZX_I2S_PROCESS_TX_EN | ZX_I2S_PROCESS_I2S_EN); + writel_relaxed(con, base + ZX_I2S_PROCESS_CTRL); +} + +static void zx_i2s_rx_en(void __iomem *base, bool on) +{ + unsigned long con; + + con = readl_relaxed(base + ZX_I2S_PROCESS_CTRL); + if (on) + con |= ZX_I2S_PROCESS_RX_EN | ZX_I2S_PROCESS_I2S_EN; + else + con &= ~(ZX_I2S_PROCESS_RX_EN | ZX_I2S_PROCESS_I2S_EN); + writel_relaxed(con, base + ZX_I2S_PROCESS_CTRL); +} + +static void zx_i2s_tx_dma_en(void __iomem *base, bool on) +{ + unsigned long con; + + con = readl_relaxed(base + ZX_I2S_FIFO_CTRL); + con |= ZX_I2S_FIFO_CTRL_TX_RST | (I2S_DEAGULT_FIFO_THRES << 8); + if (on) + con |= ZX_I2S_FIFO_CTRL_TX_DMA_EN; + else + con &= ~ZX_I2S_FIFO_CTRL_TX_DMA_EN; + writel_relaxed(con, base + ZX_I2S_FIFO_CTRL); +} + +static void zx_i2s_rx_dma_en(void __iomem *base, bool on) +{ + unsigned long con; + + con = readl_relaxed(base + ZX_I2S_FIFO_CTRL); + con |= ZX_I2S_FIFO_CTRL_RX_RST | (I2S_DEAGULT_FIFO_THRES << 16); + if (on) + con |= ZX_I2S_FIFO_CTRL_RX_DMA_EN; + else + con &= ~ZX_I2S_FIFO_CTRL_RX_DMA_EN; + writel_relaxed(con, base + ZX_I2S_FIFO_CTRL); +} + +#define ZX_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000| \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + +#define ZX_I2S_FMTBIT \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static int zx_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct zx_i2s_info *zx_i2s = dev_get_drvdata(dai->dev); + + snd_soc_dai_set_drvdata(dai, zx_i2s); + zx_i2s->dma_playback.addr = zx_i2s->mapbase + ZX_I2S_DATA; + zx_i2s->dma_playback.maxburst = 16; + zx_i2s->dma_capture.addr = zx_i2s->mapbase + ZX_I2S_DATA; + zx_i2s->dma_capture.maxburst = 16; + snd_soc_dai_init_dma_data(dai, &zx_i2s->dma_playback, + &zx_i2s->dma_capture); + return 0; +} + +static int zx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct zx_i2s_info *i2s = snd_soc_dai_get_drvdata(cpu_dai); + unsigned long con; + + con = readl_relaxed(i2s->reg_base + ZX_I2S_TIMING_CTRL); + con &= ~(ZX_I2S_TIMING_TIMING_MASK | ZX_I2S_TIMING_ALIGN_MASK | + ZX_I2S_TIMING_TEAK_MASK | ZX_I2S_TIMING_SYNC_MASK | + ZX_I2S_TIMING_MS_MASK); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + con |= (ZX_I2S_TIMING_I2S | ZX_I2S_TIMING_STD_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + con |= (ZX_I2S_TIMING_I2S | ZX_I2S_TIMING_MSB_JUSTIF); + break; + case SND_SOC_DAIFMT_RIGHT_J: + con |= (ZX_I2S_TIMING_I2S | ZX_I2S_TIMING_LSB_JUSTIF); + break; + default: + dev_err(cpu_dai->dev, "Unknwon i2s timeing\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s->master = 1; + con |= ZX_I2S_TIMING_MAST; + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s->master = 0; + con |= ZX_I2S_TIMING_SLAVE; + break; + default: + dev_err(cpu_dai->dev, "Unknwon master/slave format\n"); + return -EINVAL; + } + + writel_relaxed(con, i2s->reg_base + ZX_I2S_TIMING_CTRL); + return 0; +} + +static int zx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct zx_i2s_info *i2s = snd_soc_dai_get_drvdata(socdai); + struct snd_dmaengine_dai_dma_data *dma_data; + unsigned int lane, ch_num, len, ret = 0; + unsigned long con, format; + unsigned long chn_cfg; + + dma_data = snd_soc_dai_get_dma_data(socdai, substream); + dma_data->addr_width = params_width(params) >> 3; + + con = readl_relaxed(i2s->reg_base + ZX_I2S_TIMING_CTRL); + con &= ~(ZX_I2S_TIMING_TS_WIDTH_MASK | ZX_I2S_TIMING_DATA_SIZE_MASK | + ZX_I2S_TIMING_LANE_MASK | ZX_I2S_TIMING_CHN_MASK | + ZX_I2S_TIMING_TSCFG_MASK); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + format = 0; + len = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + format = 1; + len = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + format = 2; + len = 32; + break; + default: + dev_err(socdai->dev, "Unknwon data format\n"); + return -EINVAL; + } + con |= ZX_I2S_TIMING_TS_WIDTH(len) | ZX_I2S_TIMING_DATA_SIZE(len); + + ch_num = params_channels(params); + switch (ch_num) { + case 1: + lane = 1; + chn_cfg = 2; + break; + case 2: + case 4: + case 6: + case 8: + lane = ch_num / 2; + chn_cfg = 3; + break; + default: + dev_err(socdai->dev, "Not support channel num %d\n", ch_num); + return -EINVAL; + } + con |= ZX_I2S_TIMING_LANE(lane); + con |= ZX_I2S_TIMING_TSCFG(chn_cfg); + con |= ZX_I2S_TIMING_CHN(ch_num); + writel_relaxed(con, i2s->reg_base + ZX_I2S_TIMING_CTRL); + + if (i2s->master) + ret = clk_set_rate(i2s->dai_clk, + params_rate(params) * ch_num * CLK_RAT); + return ret; +} + +static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct zx_i2s_info *zx_i2s = dev_get_drvdata(dai->dev); + int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (capture) + zx_i2s_rx_dma_en(zx_i2s->reg_base, true); + else + zx_i2s_tx_dma_en(zx_i2s->reg_base, true); + /* fall thru */ + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (capture) + zx_i2s_rx_en(zx_i2s->reg_base, true); + else + zx_i2s_tx_en(zx_i2s->reg_base, true); + break; + case SNDRV_PCM_TRIGGER_STOP: + if (capture) + zx_i2s_rx_dma_en(zx_i2s->reg_base, false); + else + zx_i2s_tx_dma_en(zx_i2s->reg_base, false); + /* fall thru */ + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (capture) + zx_i2s_rx_en(zx_i2s->reg_base, false); + else + zx_i2s_tx_en(zx_i2s->reg_base, false); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int zx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct zx_i2s_info *zx_i2s = dev_get_drvdata(dai->dev); + + return clk_prepare_enable(zx_i2s->dai_clk); +} + +static void zx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct zx_i2s_info *zx_i2s = dev_get_drvdata(dai->dev); + + clk_disable_unprepare(zx_i2s->dai_clk); +} + +static struct snd_soc_dai_ops zx_i2s_dai_ops = { + .trigger = zx_i2s_trigger, + .hw_params = zx_i2s_hw_params, + .set_fmt = zx_i2s_set_fmt, + .startup = zx_i2s_startup, + .shutdown = zx_i2s_shutdown, +}; + +static const struct snd_soc_component_driver zx_i2s_component = { + .name = "zx-i2s", +}; + +struct snd_soc_dai_driver zx_i2s_dai = { + .name = "zx-i2s-dai", + .id = 0, + .probe = zx_i2s_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 8, + .rates = ZX_I2S_RATES, + .formats = ZX_I2S_FMTBIT, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ZX_I2S_RATES, + .formats = ZX_I2S_FMTBIT, + }, + .ops = &zx_i2s_dai_ops, +}; + +static int zx_i2s_probe(struct platform_device *pdev) +{ + struct resource *res; + struct zx_i2s_info *zx_i2s; + int ret; + + zx_i2s = kzalloc(sizeof(*zx_i2s), GFP_KERNEL); + if (!zx_i2s) + return -ENOMEM; + + zx_i2s->dai_clk = devm_clk_get(&pdev->dev, "tx"); + if (IS_ERR(zx_i2s->dai_clk)) { + dev_err(&pdev->dev, "Fail to get clk\n"); + return PTR_ERR(zx_i2s->dai_clk); + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + zx_i2s->mapbase = res->start; + zx_i2s->reg_base = devm_ioremap_resource(&pdev->dev, res); + if (!zx_i2s->reg_base) { + dev_err(&pdev->dev, "ioremap failed!\n"); + return -EIO; + } + + writel_relaxed(0, zx_i2s->reg_base + ZX_I2S_FIFO_CTRL); + platform_set_drvdata(pdev, zx_i2s); + + ret = snd_soc_register_component(&pdev->dev, &zx_i2s_component, + &zx_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register DAI failed: %d\n", ret); + return ret; + } + + ret = zx_pcm_platform_register(&pdev->dev); + if (ret) + dev_err(&pdev->dev, "Register platform PCM failed: %d\n", ret); + + return ret; +} + +static const struct of_device_id zx_i2s_dt_ids[] = { + { .compatible = "zte,zx296702-i2s", }, + {} +}; +MODULE_DEVICE_TABLE(of, zx_i2s_dt_ids); + +static struct platform_driver i2s_driver = { + .probe = zx_i2s_probe, + .driver = { + .name = "zx-i2s", + .owner = THIS_MODULE, + .of_match_table = zx_i2s_dt_ids, + }, +}; + +module_platform_driver(i2s_driver); + +MODULE_AUTHOR("Jun Nie <jun.nie(a)linaro.org>"); +MODULE_DESCRIPTION("ZTE I2S SoC DAI"); +MODULE_LICENSE("GPL"); -- 1.9.1

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Re: [alsa-devel] getting the sound to work on my asus m6s00v
by Raymond Yau 23 May '15

23 May '15

> > I have done a couple of things > First I left the laptop off with no power connected; > This made the laptop awake in the non functional state regarding the soundcard. > I then used this opportunity to update to the latest kernel. > > With the latest kernel, the sound is still not working. > Unmuting the S/PDIF in alsamixer turn on red light in the headphone jack. > > also, using 3stack-digout option doesn't bring back sound. > > This is the output from the alsa information tool > http://www.alsa-project.org/db/?f=df46a9ee01e04dde11b194e0278c9f22594540d0 > > after updating this is the output from various commands > > [ > Nothing plugged in: > ./hdajacksensetest -a > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Internal Mic): present = Yes > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Not connected): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Internal SPDIF Out, Mobile-In): present = No > Pin 0x1f (Not connected): present = No > > ./hda-verb /dev/snd/hwC0D0 0x1{4,5,8} SET_PIN_SENSE 0 nid > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > > nid = 0x18, verb = 0x709, param = 0x0 > value = 0x0 > > ./hda-verb /dev/snd/hwC0D0 0x1{4,5,8} GET_PIN_SENSE 0 nid > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x80001f41 > > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x80001f41 > > nid = 0x18, verb = 0xf09, param = 0x0 > value = 0x7fffffff > ] > > [ > With headphone plugged in > ./hdajacksensetest/hdajacksensetest -a > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = Yes > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Internal Mic): present = Yes > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Not connected): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Internal SPDIF Out, Mobile-In): present = No > Pin 0x1f (Not connected): present = No > > // and also ecact same output for hda-verb as with nothing plugged in > ] > > [ > with Mic plugged in, the following items changed > > ./hdajacksensetest/hdajacksensetest -a > Pin 0x14 (Green Headphone, Rear side): present = Yes > Pin 0x15 (Not connected): present = Yes > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Internal Mic): present = Yes > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Not connected): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Internal SPDIF Out, Mobile-In): present = No > Pin 0x1f (Not connected): present = No > > ./hda-verb /dev/snd/hwC0D0 0x15 GET_PIN_SENSE 0 nid > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > > ./hda-verb /dev/snd/hwC0D0 0x18 GET_PIN_SENSE 0 nid > nid = 0x18, verb = 0xf09, param = 0x0 > value = 0x80001f41 Seem impedance measurement is not correct, mic should have higher impedance than headphone https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… You can disable it by hint trigger_sense=0 - trigger_sense (bool): indicates that the jack detection needs the explicit call of AC_VERB_SET_PIN_SENSE verb Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Master Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Mic Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x0121411f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x0c To get back headphone phantom jack if HP is really shared with speaker Disable hp mic by hint hp_mic_detect=0 - hp_mic_detect (bool): enable/disable the hp/mic shared input for a single built-in mic case; default true

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[alsa-devel] [PATCH v2 0/8] ASoC: Intel: refactor common SST IPC handling
by Vinod Koul 22 May '15

22 May '15

This series refactors the common SST IPC code to remove some hardcoded assumption which are no longer valid for future platforms like Skylake, These are made configurable and right configuration applied for respective drivers. The changes are mailbox sizes and dsp busy checks. changes in v2: - fix the mem allocation for IPC mailbox Subhransu S. Prusty (8): ASoC: Intel: Create an ops to check for DSP busy ASoC: Intel: Move the busy check to ops for Baytrail ASoC: Intel: Move the busy check to ops for HSW ASoC: Intel: Remove the direct register reference from common ipc ASoC: Intel: Allow to configure max size for mailbox data ASoC: Intel: Initialize max mailbox size for baytrail ASoC: Intel: Initialize max mailbox size for haswell ASoC: Intel: Allocate for the mailbox with max size sound/soc/intel/baytrail/sst-baytrail-ipc.c | 11 ++++++++++ sound/soc/intel/common/sst-ipc.c | 34 +++++++++++++++++++++++++---- sound/soc/intel/common/sst-ipc.h | 7 ++++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 12 ++++++++++ 4 files changed, 58 insertions(+), 6 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 00/10] ASoC: qcom: add support to apq8016 audio
by Srinivas Kandagatla 22 May '15

22 May '15

Hi Mark, Am resending the the patches with rebased on top of topic/qcom branch. All these patches are tested and Acked. Thankyou for reviewing the v2 patches, here is the v3 patchset after incorporating review comments and testing on Storm Board. This patchset adds apq8016 audio support into lpass driver. Existing Lpass driver can not be used as-it-is for apq8016 as it contains code specific to ipq806x. Also the driver only supports single i2s port, single dma channel and single bitclk control. APQ8016 has 4 MI2S( Primary, Secondary, Tertiary, Quaternary) which can be routed to internal wcd codec or external codecs. This routing is controlled by 2 mux registers. This patch series firstly re-organizes the lpass driver such that the SOC specific bits are moved away from the driver. And secondly the SOC specifics are now passed as lpass variant data which would include various register offsets, dma channel allocations and SOC specific clock handling. Finally the last few patchs add apq8016 lpass and machine driver. All these patches are tested for HDMI audio via adv7533 bridge and Analog audio on APQ8016-SBC, msm8916-mtp boards and Kenneth tested this patchset on ipq806x Storm board too. Changes since v2(https://lkml.org/lkml/2015/5/16/236) - dropped 3 patches which are already applied by Mark. - rebased on top of topic/qcom branch. - changed dev_err to dev_warn spotted by Kenneth. - fixed typo spotted by kenneth. Changes since v1(https://lwn.net/Articles/644266/) - Fixed Kconfig dependencies spotted by Kenneth - Fixed compilation error on ipq806x spotted by Kenneth - Fixed clk error messages spotted by Kenneth Changes since RFC(https://lwn.net/Articles/642661/) - droped cleanup patches, as they are already applied by Mark. - prefixed shared functions with asoc_qcom_* as suggested by Kenneth - made LPASS driver depended on OF. - removed ipq806x specific bits from ipaif_reg.h - removed no_osr clk flag as suggested by Mark and Kenneth - fixed irq return values spotted by Kenneth - renamed the machine file to apq8016-sbc.c as suggested by Kenneth. - use dai->driver->id instead of dai->id as it breaks on Storm board. --srini Srinivas Kandagatla (10): ASoC: qcom: make lpass driver depend on OF ASoC: qcom: support bitclk and osrclk per i2s port ASoC: qcom: make osr clock optional ASoC: qcom: add dma channel control offset to variant data ASoC: qcom: Add ability to handle interrupts per dma channel ASoC: qcom: add bit map to track static dma channel allocations ASoC: qcom: Add apq8016 lpass driver support ASoC: qcom: add apq8016 sound card support ASoC: qcom: Document apq8016 bindings. ASoC: qcom: document apq8016 sbc machine driver bindings .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 61 ++++++ .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 13 +- include/dt-bindings/sound/apq8016-lpass.h | 9 + sound/soc/qcom/Kconfig | 17 ++ sound/soc/qcom/Makefile | 4 + sound/soc/qcom/apq8016_sbc.c | 215 ++++++++++++++++++ sound/soc/qcom/lpass-apq8016.c | 242 +++++++++++++++++++++ sound/soc/qcom/lpass-cpu.c | 77 +++++-- sound/soc/qcom/lpass-platform.c | 96 +++++--- sound/soc/qcom/lpass.h | 21 +- 10 files changed, 692 insertions(+), 63 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt create mode 100644 include/dt-bindings/sound/apq8016-lpass.h create mode 100644 sound/soc/qcom/apq8016_sbc.c create mode 100644 sound/soc/qcom/lpass-apq8016.c -- 1.9.1

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[alsa-devel] [PATCH 0/4 v2] ASoC: rsnd: fixup interrupt / spin lock issue
by Kuninori Morimoto 22 May '15

22 May '15

Hi Mark, Vinod Cc Laurent, Hiep-san These are v2 of fixup unknown interrupt issue patch which was reported by Hiep-san in v4.1-rcX kernel. It was not happened in v4.0 because ARM platform side couldn't use DMA transfer. We tested these same as v1 patches, and confirmed these solved problem Hiep-san, can you please re-test these patches ? Kuninori Morimoto (4): 1) dmaengine: rcar-dmac: fixup spinlock in rcar-dmac 2) ASoC: rsnd: indicate unknown HW start 3) ASoC: rsnd: add rsnd_dai_stream_quit() 4) ASoC: rsnd: spin lock for interrupt handler drivers/dma/sh/rcar-dmac.c | 55 +++++++++++++++++++++++++++++++------------------------ sound/soc/sh/rcar/core.c | 25 ++++++++++++++++++------- sound/soc/sh/rcar/rsnd.h | 3 +-- sound/soc/sh/rcar/src.c | 11 ++++++++--- sound/soc/sh/rcar/ssi.c | 18 ++++++++++++++---- 5 files changed, 72 insertions(+), 40 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe linux-sh" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

4 15

[alsa-devel] [PATCH 0/2 v2] ASoC: DPCM can care FE/BE merged format
by Kuninori Morimoto 22 May '15

22 May '15

Hi Mark These are v2 patches of new .dpcm_merged_foramt for merging FE/BE formats. It will be no sound if DPCM were below style, and user selects S24_LE. FE: S16_LE/S24_LE BE: S16_LE DPCM will selects FE/BE merged formats (= S16_LE here) if FE has .dpcm_merged_foramt. Kuninori Morimoto (2): ASoC: soc-pcm: DPCM cares BE format ASoC: rsnd: rsrc-card uses FE/BE merged format when DPCM include/sound/soc.h | 3 +++ sound/soc/sh/rcar/rsrc-card.c | 1 + sound/soc/soc-pcm.c | 47 ++++++++++++++++++++++++++++++++++++++++++----- 3 files changed, 46 insertions(+), 5 deletions(-) Best regards --- Kuninori Morimoto _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 5

[alsa-devel] [PATCH] ASoC: codecs-ac97: make selectable in config
by Maciej S. Szmigiero 22 May '15

22 May '15

Make generic ASoC AC'97 CODEC selectable in config. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers but which are not directly selectable in config themselves (for example DT based ones). Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AC97_CODEC select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X_SPI if SPI_MASTER select SND_SOC_AD193X_I2C if I2C @@ -212,8 +212,9 @@ config SND_SOC_AB8500_CODEC tristate config SND_SOC_AC97_CODEC - tristate + tristate "Build generic ASoC AC97 CODEC driver" select SND_AC97_CODEC + select SND_SOC_AC97_BUS config SND_SOC_AD1836 tristate

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Re: [alsa-devel] [PATCH v2 2/3] ASoC: zx: Add zx296702 SPDIF support
by Mark Brown 22 May '15

22 May '15

On Wed, May 06, 2015 at 08:38:04PM +0800, Jun Nie wrote: > Add driver for zx296702 SPDIF controller This all seems fine apart from the fact that you're adding a new DT binding with no binding documentation - all new DT binndings require binnding documentation.

1 0

Re: [alsa-devel] [PATCH v2 1/3] ASoC: zx: Add ZTE zx296702 pcm support
by Mark Brown 22 May '15

22 May '15

On Wed, May 06, 2015 at 08:38:03PM +0800, Jun Nie wrote: > +static const struct snd_pcm_hardware snd_zx_hardware = { > + .info = SNDRV_PCM_INFO_MMAP | > + SNDRV_PCM_INFO_MMAP_VALID | > + SNDRV_PCM_INFO_INTERLEAVED | > + SNDRV_PCM_INFO_PAUSE | > + SNDRV_PCM_INFO_RESUME | > + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, > + .period_bytes_min = 32, > + .period_bytes_max = 16 * 1024, > + .periods_min = 2, > + .periods_max = 32, > + .buffer_bytes_max = 64 * 1024, > +}; As I said last time I'm not sure why you are manually defining these - the capabilities of the DMA controller should be being enumerated from the DMA controller driver. See dmaengine_pcm_set_runtime_hwparams().

1 0

[alsa-devel] how to ask you to write a new ALSA driver
by jean-noël chardron 22 May '15

22 May '15

hello, Last week I got a Komplet Audio 6 from Native intruments. The driver exist in ALSA driver for this king of interface audio (snd-usb-audio) (thanks for the team who write this) but for the model where Id Vendor is 0x17cc and Id product is 0x1000. The KA6 that I got has not the same ID product (0x1001) therefore I don't have any device capture even with the driver snd-usb-audio. The challenge is to write the modification for this new model. Who could do that ? If necessary I could send log, compile kernel source and so on. thanks, Jean-noel

2 3

[alsa-devel] [PATCH] ASoC: qcom: remove incorrect dependencies
by Arnd Bergmann 21 May '15

21 May '15

Compile-tests show a warning for the newly added SND_SOC_STORM symbol: warning: (SND_SOC_STORM) selects SND_SOC_LPASS_CPU which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && SND_SOC_QCOM) The problem is that it can be selected for COMPILE_TEST on non-QCOM builds, but the symbols it selects have a dependency. Dropping the dependencies makes it work without warnings and no other side-effects, because these are not user-visible. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: f380dd3f3cd ("ASoC: qcom: Add ability to build QCOM drivers") diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 5f58e4f1bca9..b07f183fc47f 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -6,12 +6,10 @@ config SND_SOC_QCOM config SND_SOC_LPASS_CPU tristate - depends on SND_SOC_QCOM select REGMAP_MMIO config SND_SOC_LPASS_PLATFORM tristate - depends on SND_SOC_QCOM select REGMAP_MMIO config SND_SOC_STORM

3 3

[alsa-devel] [PATCH] ASoC: skip legacy dai naming if dai driver has all the information
by Srinivas Kandagatla 21 May '15

21 May '15

Original issue is that the id field in the dai is not same as the id in dai_driver when dai driver count == 1. This is due to the legacy dai naming check, which could possibly cause issues if the audio drivers written in assumption that dai->id would be always equal to dai_driver->id. This assumption is true only if the dai driver count is greater than 1, and false if dai driver count is 1. On Qcom Lpass driver we hit such issue while adding support to apq8016. The code path which falls back to legacy naming for cases where num_dai == 1 does not check if there is any valid information in the dai_driver. This patch fixes that by checking if the dai_driver has valid id and name before falling back to legacy dai naming Although the drivers can work around this issue by only using dai->driver->id, but this patch attempts to fix the actual issue. Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2373252..7d028e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2599,7 +2599,8 @@ static int snd_soc_register_dais(struct snd_soc_component *component, * the same naming style even though those DAIs are not * component-less anymore. */ - if (count == 1 && legacy_dai_naming) { + if (count == 1 && legacy_dai_naming && + (dai_drv[i].id == 0 || dai_drv[i].name == NULL)) { dai->name = fmt_single_name(dev, &dai->id); } else { dai->name = fmt_multiple_name(dev, &dai_drv[i]); -- 1.9.1

3 2

[alsa-devel] Assert in pcm_params.c
by Alan Horstmann 21 May '15

21 May '15

Hi, The following assert is being hit in pcm_params.c: "pcm_params.c:2249: snd1_pcm_hw_params_slave: Assertion `err >= 0' failed." This is from a developer using Portaudio in his own application; I have yet to ascertain full details, but it is trying successive sample rates to see which are valid. Does this point to a flaw in the configuration or possibly a bug in Alsa-lib or can it occur just as a result of misuse of the Alsa API? Regards Alan

3 21

[alsa-devel] [RFC PATCH 00/14] ASoC: qcom: add support to apq8016 audio
by Srinivas Kandagatla 21 May '15

21 May '15

Hi All, This patchset adds apq8016 audio support into lpass driver. Existing Lpass driver can not be used as-it-is for apq8016 as it contains code specific to ipq806x. Also the driver only supports single i2s port, single dma channel and single bitclk control. APQ8016 has 4 MI2S( Primary, Secondary, Tertiary, Quaternary) which can be routed to internal wcd codec or external codecs. This routing is controlled by 2 mux registers. This patch series firstly re-organizes the lpass driver such that the SOC specific bits are moved away from the driver. And secondly the SOC specifics are now passed as lpass variant data which would include various register offsets, dma channel allocations and SOC specific clock handling. Finally the patchset add apq8016 lpass and machine driver. This patchset also has two trivial cleanup patches which are to do with redundant checks and removing unnecessary header files. All these patches are tested for HDMI audio via adv7533 bridge and Analog audio on APQ8016-SBC and msm8916-mtp boards. I dont have access to ipq806x boards to test these patches. This is very first version of the patches which was developed with very mimimal/no access to IP documentation. I would like to get your opinon on the over all approch. Kenneth/Patrick, Could you please try these patches on storm board? Thanks, srini Srinivas Kandagatla (14): ASoC: qcom: Remove redundant error check. ASoC: qcom: remove unnecessary header files ASoC: qcom: move ipq806x specific bits out of lpass driver. ASoC: qcom: remove hardcoded i2s port number ASoC: qcom: remove hardcoded dma channel ASoC: qcom: support bitclk and osrclk per i2s port ASoC: qcom: add no osr clk flag to lpass variant ASoC: qcom: add dma channel control offset to variant data ASoC: qcom: Add ability to handle interrupts per dma channel ASoC: qcom: add bit map to track static dma channel allocations ASoC: qcom: Add apq8016 lpass driver support ASoC: qcom: add apq8016 sound card support ASoC: qcom: Document apq8016 bindings. ASoC: qcom: document apq8016 machine driver bindings .../bindings/sound/qcom,apq8016-machine.txt | 61 ++++++ .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 13 +- sound/soc/qcom/Kconfig | 25 ++- sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8016.c | 214 ++++++++++++++++++ sound/soc/qcom/lpass-apq8016.c | 243 +++++++++++++++++++++ sound/soc/qcom/lpass-cpu.c | 234 ++++++++++---------- sound/soc/qcom/lpass-ipq806x.c | 95 ++++++++ sound/soc/qcom/lpass-lpaif-ipq806x.h | 172 --------------- sound/soc/qcom/lpass-lpaif-reg.h | 167 ++++++++++++++ sound/soc/qcom/lpass-platform.c | 197 +++++++++++------ sound/soc/qcom/lpass.h | 54 ++++- 12 files changed, 1121 insertions(+), 359 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-machine.txt create mode 100644 sound/soc/qcom/apq8016.c create mode 100644 sound/soc/qcom/lpass-apq8016.c create mode 100644 sound/soc/qcom/lpass-ipq806x.c delete mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass-lpaif-reg.h -- 1.9.1

4 87

[alsa-devel] prealloc_buffer_size and buffer_bytes_max
by Peter Rosin 21 May '15

21 May '15

Hi! I got my hopes up when I read the commit message for c14e2591bf54 ASoC: atmel-pcm-dma: increase buffer_bytes_max. Quoting it: atmel-pcm-dma is not limited to a buffer size of 64kB like atmel-pcm-pdc. Increase buffer_bytes_max to 512kB to allow for higher bit rates (i.e. 32bps at 192kHz) to work correctly. By default, keep the prealloc at 64kB. However, as I (think I) request a bigger buffer it still caps out at 64kB. I'm using the latency argument of snd_pcm_set_params to control the buffer: snd_pcm_set_params(pcm, SND_PCM_FORMAT_S32_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 2, /* channels */ 250000, /* rate */ 0, /* do not resample */ 90000); /* latency in us */ But snd_pcm_hw_params_get_buffer_size "only" returns 8192 frames (64kB) even if I request 9us * 250kHz * 2 * 4 = 180kB. If I change the prealloc from 64kB to 256kB I get a bigger buffer (and it works better too!). Admittedly I backported this patch on top of the linux-3.18-at91 branch from the Atmel git repo, so there might be some support missing that has gone in after 3.18? Or have I completely misunderstood, and these are unrelated buffers? Any insight in how I can get a big enough buffer without hacking the prealloc is appreciated! Cheers, Peter

2 3

[alsa-devel] snd-hda-intel : issues with mic boost on ALC3236
by Simon Derr 21 May '15

21 May '15

Hello, I'm having issues with the internal microphone on an Asus R553 (or S551) laptop. With the boost set to zero the microphone works but the level of the recording is very low. If I try another boost setting I don't get any signal anymore. I have uploaded alsa-info data here : http://www.alsa-project.org/db/?f=55b7fb0b4998757d099addae0a411e6a7cebc079 This is on a Linux 4.0 kernel. Any help is welcome, Regards, Simon

4 13

[alsa-devel] [REGRESSION] 4.1-rc4: Sound muted after boot
by Martin Steigerwald 21 May '15

21 May '15

Hello! On ThinkPad T520 (Intel Sandybridge) with 4.14-rc4 + two BTRFS patches and one Intel P-State debug patch I have no sound output on internal Intel HD audio codec. According to ThinkPad T520 LED with striked-through speaker the loud speaker is in disabled state. Also KDE´s Phonon says a soundcard is missing. In alsa-mixer I see nothing as MM for muted. If I press the key next to the LED I get a notification on the desktop that loud speaker has been enabled / disabled, but the LED still is on for it being disabled. I do not hear anything in Amarok or with aplay. alsa-mixer sound volumes are 100 for master and PCM. I also tried with disabling "Auto-Mute Mod", but still no sound. When I reboot and in BIOS or GRUB press the key then the LED toggles state. When I reboot into Linux 4.0.1 sound is working fine and loud speaker disabled LED is toggled off, so I bet this is a regression 4.1-rc4 compared to 4.0.1. The following is from non-working 4.1-rc4 kernel: merkaba:~> lspci -s 00:1b.0 -nnv 00:1b.0 Audio device [0403]: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller [8086:1c20] (rev 04) Subsystem: Lenovo Device [17aa:21cf] Flags: bus master, fast devsel, latency 0, IRQ 29 Memory at f2520000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel Capabilities: [130] Root Complex Link Kernel driver in use: snd_hda_intel merkaba:~> cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf2520000 irq 29 merkaba:~> cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k4.1.0-rc4-tp520-btrfstrim- pstatetrace+. merkaba:~> cat /proc/asound/devices 2: [ 0] : control 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 5: [ 0- 3]: digital audio playback 6: [ 0- 7]: digital audio playback 7: [ 0- 8]: digital audio playback 8: [ 0- 0]: hardware dependent 9: [ 0- 1]: hardware dependent 10: [ 0- 3]: hardware dependent 33: : timer merkaba:~> merkaba:~> aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server phonon HDA Intel PCH mit Mixer default:CARD=PCH HDA Intel PCH, CX20590 Analog Default Audio Device sysdefault:CARD=PCH HDA Intel PCH, CX20590 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=PCH,DEV=0 HDA Intel PCH, HDMI 0 HDMI Audio Output hdmi:CARD=PCH,DEV=1 HDA Intel PCH, HDMI 1 HDMI Audio Output hdmi:CARD=PCH,DEV=2 HDA Intel PCH, HDMI 2 HDMI Audio Output dmix:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog Direct sample mixing device dmix:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample mixing device dmix:CARD=PCH,DEV=7 HDA Intel PCH, HDMI 1 Direct sample mixing device dmix:CARD=PCH,DEV=8 HDA Intel PCH, HDMI 2 Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog Direct sample snooping device dsnoop:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample snooping device dsnoop:CARD=PCH,DEV=7 HDA Intel PCH, HDMI 1 Direct sample snooping device dsnoop:CARD=PCH,DEV=8 HDA Intel PCH, HDMI 2 Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog Direct hardware device without any conversions hw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct hardware device without any conversions hw:CARD=PCH,DEV=7 HDA Intel PCH, HDMI 1 Direct hardware device without any conversions hw:CARD=PCH,DEV=8 HDA Intel PCH, HDMI 2 Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog Hardware device with all software conversions plughw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Hardware device with all software conversions plughw:CARD=PCH,DEV=7 HDA Intel PCH, HDMI 1 Hardware device with all software conversions plughw:CARD=PCH,DEV=8 HDA Intel PCH, HDMI 2 Hardware device with all software conversions Ciao, -- Martin 'Helios' Steigerwald - http://www.Lichtvoll.de GPG: 03B0 0D6C 0040 0710 4AFA B82F 991B EAAC A599 84C7

2 9

Re: [alsa-devel] [PATCH 1/4] ASoC: rockchip: add rockchip machine driver
by Mark Brown 21 May '15

21 May '15

On Wed, May 13, 2015 at 09:23:01PM +0800, zhengxing wrote: > On 2015年05月13日 03:22, Mark Brown wrote: > >Is it not possible to extend simple card to handle your use cases? > >Given the very generic naming and the fact that things like jack > >detection and so on should depend on the CODEC and board usually rather > >than on the SoC it doesn't sound like this is Rockchip specific. > >This also looks like you're reimplementing some device model enumeration > >stuff which probably shouldn't be happening but let's understand the > >problem you're trying to solve here before going too far into the code. > Because we are trying to bring rt5650 in the project, so we intend to > describe supported codecs with DTS via only a rockchip machine driver file, > others remain pre-implement(like max98090 / rt5645 that vendor machine > driver). I don't undertand what you're saying here, sorry - why is this not just a case of writing multiple machine drivers? > I communicated with Dylan, and he told me that the jack detection is an > issue in the simple-card, and suggested we are better to send them at > present. But what are these issues?

3 6

[alsa-devel] [PATCH] ASoC: mop500_ab8500: Add a NULL pointer check in mop500_ab8500_machine_init()
by Rajan Vaja 21 May '15

21 May '15

Avoid possible crash (NULL pointer dereference) by making sure that dem_kzalloc() is successful. Signed-off-by: Rajan Vaja <rajan.vaja(a)gmail.com> --- sound/soc/ux500/mop500_ab8500.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index b81a7a4c..85d810d 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -372,6 +372,10 @@ int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) /* Create driver private-data struct */ drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata), GFP_KERNEL); + + if (!drvdata) + return -ENOMEM; + snd_soc_card_set_drvdata(rtd->card, drvdata); /* Setup clocks */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: jack: Remove MODULE_*() macros
by Takashi Iwai 21 May '15

21 May '15

The jack interface is statically included in sound core. Having doubly module information is rather confusing. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/jack.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/core/jack.c b/sound/core/jack.c index eb66327b5f1c..7237acbdcbbc 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -372,7 +372,3 @@ void snd_jack_report(struct snd_jack *jack, int status) } EXPORT_SYMBOL(snd_jack_report); - -MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -MODULE_DESCRIPTION("Jack detection support for ALSA"); -MODULE_LICENSE("GPL"); -- 2.4.1

1 0

[alsa-devel] [PATCH] ASoC: rx51: use flags argument of devm_gpiod_get to set direction
by Uwe Kleine-König 20 May '15

20 May '15

Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Use this to simplify the driver. Furthermore this is one caller less that stops us making the flags argument to gpiod_get*() mandatory. Fixes: 386e81ab3b4d ("ASoC: omap: rx51: get GPIO numbers via gpiod API") Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/omap/rx51.c | 30 ++++++++---------------------- 1 file changed, 8 insertions(+), 22 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index c2ddf0fbfa28..fded99362d39 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -455,50 +455,36 @@ static int rx51_soc_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, pdata); pdata->tvout_selection_gpio = devm_gpiod_get(card->dev, - "tvout-selection"); + "tvout-selection", + GPIOD_OUT_LOW); if (IS_ERR(pdata->tvout_selection_gpio)) { dev_err(card->dev, "could not get tvout selection gpio\n"); return PTR_ERR(pdata->tvout_selection_gpio); } - err = gpiod_direction_output(pdata->tvout_selection_gpio, 0); - if (err) { - dev_err(card->dev, "could not setup tvout selection gpio\n"); - return err; - } - pdata->jack_detection_gpio = devm_gpiod_get(card->dev, - "jack-detection"); + "jack-detection", + GPIOD_ASIS); if (IS_ERR(pdata->jack_detection_gpio)) { dev_err(card->dev, "could not get jack detection gpio\n"); return PTR_ERR(pdata->jack_detection_gpio); } - pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch"); + pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch", + GPIOD_OUT_HIGH); if (IS_ERR(pdata->eci_sw_gpio)) { dev_err(card->dev, "could not get eci switch gpio\n"); return PTR_ERR(pdata->eci_sw_gpio); } - err = gpiod_direction_output(pdata->eci_sw_gpio, 1); - if (err) { - dev_err(card->dev, "could not setup eci switch gpio\n"); - return err; - } - pdata->speaker_amp_gpio = devm_gpiod_get(card->dev, - "speaker-amplifier"); + "speaker-amplifier", + GPIOD_OUT_LOW); if (IS_ERR(pdata->speaker_amp_gpio)) { dev_err(card->dev, "could not get speaker enable gpio\n"); return PTR_ERR(pdata->speaker_amp_gpio); } - err = gpiod_direction_output(pdata->speaker_amp_gpio, 0); - if (err) { - dev_err(card->dev, "could not setup speaker enable gpio\n"); - return err; - } - err = devm_snd_soc_register_card(card->dev, card); if (err) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err); -- 2.1.4

3 4

[alsa-devel] [PATCH] SoC: max98357a: use flags argument of devm_gpiod_get to set direction
by Uwe Kleine-König 20 May '15

20 May '15

Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Use this to simplify the driver. Furthermore this is one caller less that stops us making the flags argument to gpiod_get*() mandatory. Fixes: af5adf129369 ("ASoC: max98357a: Add MAX98357A codec driver") Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/max98357a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index bf3e933ee895..3a2fda08a893 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -60,13 +60,12 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec) { struct gpio_desc *sdmode; - sdmode = devm_gpiod_get(codec->dev, "sdmode"); + sdmode = devm_gpiod_get(codec->dev, "sdmode", GPIOD_OUT_LOW); if (IS_ERR(sdmode)) { dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n", __func__, PTR_ERR(sdmode)); return PTR_ERR(sdmode); } - gpiod_direction_output(sdmode, 0); snd_soc_codec_set_drvdata(codec, sdmode); return 0; -- 2.1.4

3 2

[alsa-devel] [PATCH] SoC: sta32x: use devm_gpiod_get_optional for optional reset gpio
by Uwe Kleine-König 20 May '15

20 May '15

Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Also there is a variant to find optional gpios that returns NULL if there is no gpio instead of -ENOENT. Make use of both features to simplify the driver. This changes behaviour if gpiod_get returns -ENOSYS which is the case if CONFIG_GPIOLIB is not enabled. This is a good change because without GPIOLIB there is no way to determine if the reset gpio is specified in the device tree. And if it is it must be handled, so erroring out is the right thing to do. Furthermore this is one caller less that stops us making the flags argument to gpiod_get*() mandatory. Fixes: b66a29808e1f ("ASoC: sta32x: make sta32x a gpio consumer for the reset GPIO") Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/sta32x.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 007a0e3bc273..0111baf9a5d4 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1096,16 +1096,10 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta32x->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta32x->gpiod_nreset)) { - ret = PTR_ERR(sta32x->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta32x->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta32x->gpiod_nreset, 0); - } + sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta32x->gpiod_nreset)) + return PTR_ERR(sta32x->gpiod_nreset); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency, again
by Arnd Bergmann 20 May '15

20 May '15

I tried to fix this before and submitted a working patch, but after some discussion we came up with what seemed to be a nicer solution, resulting in commit 3d4cf65e2d ("ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency"). Unfortunately, that version was incomplete, and we still get this build error: drivers/clk/clk-palmas.c:46:16: error: field 'hw' has incomplete type drivers/clk/clk-palmas.c: In function 'to_palmas_clks_info': drivers/clk/clk-palmas.c:54:74: warning: initialization from incompatible pointer type [-Winc This happens only in randconfig builds that turn on MFD_PALMAS on a platform other than OMAP2+ when COMPILE_TEST is set but COMMON_CLK is not. The new approach is only 'select COMMON_CLK_PALMAS' if we know that we are on an OMAP5 platform and MFD_PALMAS is already set. This patch has survived thousands of randconfig builds and I don't see a remaining hole in the logic. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 3d4cf65e2d ("ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency") diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 6768e4f7d7d0..30d0109703a9 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -100,12 +100,13 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST) + depends on TWL6040_CORE && SND_OMAP_SOC + depends on ARCH_OMAP4 || (SOC_OMAP5 && MFD_PALMAS) || COMPILE_TEST select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC - select COMMON_CLK_PALMAS if MFD_PALMAS + select COMMON_CLK_PALMAS if (SOC_OMAP5 && MFD_PALMAS) help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports:

3 2

Re: [alsa-devel] [PATCH v3 06/17] video: lcd: add LoCoMo LCD driver
by Jingoo Han 20 May '15

20 May '15

> LoCoMo has some special handling for TFT screens attached to Collie and > Poodle. Implement that as a separate driver. > > Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> Acked-by: Jingoo Han <jingoohan1(a)gmail.com> Best regards, Jingoo Han > --- > drivers/video/backlight/Kconfig | 10 ++ > drivers/video/backlight/Makefile | 1 + > drivers/video/backlight/locomo_lcd.c | 285 +++++++++++++++++++++++++++++++++++ > 3 files changed, 296 insertions(+) > create mode 100644 drivers/video/backlight/locomo_lcd.c

1 0

[alsa-devel] [PATCH 1/7] ASoC: alc5623: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 20 May '15

20 May '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm with snd_soc_codec_get_dapm(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Full series depends on topic/dapm --- sound/soc/codecs/alc5623.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index e92b5ae..0fc24e0 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -893,7 +893,7 @@ static int alc5623_resume(struct snd_soc_codec *codec) static int alc5623_probe(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); alc5623_reset(codec); -- 1.8.0

2 7

[alsa-devel] [patch] ASoC: wm8741: check for error returns from wm8741_set_pdata()
by Dan Carpenter 20 May '15

20 May '15

Static checkers complain that "ret" is always zero so the conditions are never true. The intention here was clearly to check for errors from wm8741_set_pdata(). Although, since wm8741_set_pdata() never returns errors, it doesn't affect runtime. Fixes: c354b54cfdf6 ('ASoC: wm8741: Add differential mono mode support') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index c065ea1..09ff01f 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -613,7 +613,7 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, return ret; } - wm8741_set_pdata(&i2c->dev, wm8741); + ret = wm8741_set_pdata(&i2c->dev, wm8741); if (ret != 0) { dev_err(&i2c->dev, "Failed to set pdata: %d\n", ret); return ret; @@ -679,7 +679,7 @@ static int wm8741_spi_probe(struct spi_device *spi) return ret; } - wm8741_set_pdata(&spi->dev, wm8741); + ret = wm8741_set_pdata(&spi->dev, wm8741); if (ret != 0) { dev_err(&spi->dev, "Failed to set pdata: %d\n", ret); return ret;

3 2

[alsa-devel] ALSA: dmix: pcm status incomplete
by Dan Hordern 20 May '15

20 May '15

I require timestamp information from the soundcard and host for our application. However, I noticed that when using "dmix:0,0" instead of "hw:0,0", the audio timestamp and delay frames are both zero. Looking in to the code, it seems that the 'snd_pcm_dmix_status' function does not populate either of these. In order to expose these, I have modified this function to get this information from the slave pcm; patch below. I was hoping for some feedback on this change. I am not sure if the delay value makes sense as yet, and whether there are consequences of using the hardware trigger timestamp etc. I am guessing the dsnoop and dshare plugins may also have this issue, though I have not checked. Thanks, Dan diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index babde6a..a4d7fc0 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -475,12 +475,12 @@ static int snd_pcm_dmix_status(snd_pcm_t *pcm, snd_pcm_status_t * status) break; } memset(status, 0, sizeof(*status)); + snd_pcm_status(dmix->spcm, status); status->state = snd_pcm_dmix_state(pcm); - status->trigger_tstamp = dmix->trigger_tstamp; - gettimestamp(&status->tstamp, pcm->tstamp_type); status->avail = snd_pcm_mmap_playback_avail(pcm); status->avail_max = status->avail > dmix->avail_max ? status->avail : dmix->avail_max; dmix->avail_max = 0; + status->delay = snd_pcm_mmap_playback_delay(pcm); return 0; }

2 1

[alsa-devel] [PATCH 0/4] ASoC: rsnd: fixup interrupt / spin lock issue
by Kuninori Morimoto 20 May '15

20 May '15

Hi Mark, Vinod Cc Laurent These patches fixup unknown interrupt issue which was reported by Hiep-san in v4.1-rcX kernel. It was not happened in v4.0 because ARM platform side couldn't use DMA transfer. --------- Unable to handle kernel NULL pointer dereference at virtual address 00000048 pgd = edfac000 [00000048] *pgd=6e0f0831, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#1] SMP ARM CPU: 0 PID: 2009 Comm: aplay Not tainted 4.1.0-rc2-dirty #4 Hardware name: Generic R8A7790 (Flattened Device Tree) ... -------- Renesas sound driver had 2 issues. [1] spin lock was unlocked un-expected timing [2] HW restart (when error timing) is called without spin lock In our environment, it was difficult to reproduce this error, because it happens in very rare timing. But I noticed it is easy to reproduce it if I added this setting. [Kernel hacking] - [Lock Debugging] - all ON And we decided this error has gone by these tests. Test1: reproduce error 1. start / stop sound playback in 100 times 2. count error 3. test above in 10 sets result: 100% error happen. In average, this error happens once during 5 playbacks. Test2: apply these fixup patces, and retry Test1 result: 0 error happen. Kuninori Morimoto (4): 1) dmaengine: rcar-dmac: fixup spinlock in rcar-dmac 2) ASoC: rsnd: indicate unknown HW start 3) ASoC: rsnd: add rsnd_dai_stream_quit() 4) ASoC: rsnd: spin lock for interrupt handler drivers/dma/sh/rcar-dmac.c | 55 +++++++++++++++++++++++++++++++------------------------ sound/soc/sh/rcar/core.c | 25 ++++++++++++++++++------- sound/soc/sh/rcar/rsnd.h | 3 +-- sound/soc/sh/rcar/src.c | 11 ++++++++--- sound/soc/sh/rcar/ssi.c | 18 ++++++++++++++---- 5 files changed, 72 insertions(+), 40 deletions(-)

3 6

[alsa-devel] [PATCH] ALSA: hda - Reduce verbs by node power-saves
by Takashi Iwai 20 May '15

20 May '15

The widget (node) power-saves restore the widget states at each transition from D3 to D0 on each node. This was added in the commit [d545a57c5f84:ALSA: hda - Sync node attributes at resume from widget power saving]. However, the test was rater false-positive; this wasn't needed for any codecs. Since the resync may take significant number of additional verbs to be executed, it's better to reduce it. Let's disable it for now again. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 788f969b1a68..1c8678775f40 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -844,8 +844,16 @@ static hda_nid_t path_power_update(struct hda_codec *codec, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, state); changed = nid; + /* all known codecs seem to be capable to handl + * widgets state even in D3, so far. + * if any new codecs need to restore the widget + * states after D0 transition, call the function + * below. + */ +#if 0 /* disabled */ if (state == AC_PWRST_D0) snd_hdac_regmap_sync_node(&codec->core, nid); +#endif } } return changed; -- 2.4.1

1 0

[alsa-devel] [PATCH] ALSA: sound/atmel/ac97c.c: remove unused variable
by Arnd Bergmann 20 May '15

20 May '15

The recently added DT support for the ac97 driver is causing a gcc warning: sound/atmel/ac97c.c: In function 'atmel_ac97c_probe_dt': sound/atmel/ac97c.c:919:29: warning: unused variable 'match' [-Wunused-variable] const struct of_device_id *match; The variable is clearly unused, so we can remove it. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index cf4cedf2b420..6dad042630d8 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -916,7 +916,6 @@ static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) { struct ac97c_platform_data *pdata; struct device_node *node = dev->of_node; - const struct of_device_id *match; if (!node) { dev_err(dev, "Device does not have associated DT data\n");

3 2

[alsa-devel] [PATCH v2] ALSA: hda - Move hda_i915.c from sound/pci/hda to sound/hda
by mengdong.lin＠intel.com 20 May '15

20 May '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The file is moved to hda core and renamed to hdac_i915.c, so can be used by both legacy HDA driver and new Skylake audio driver. - Add snd_hdac_ prefix to the public APIs. - The i915 audio component is moved to core bus and dynamically allocated. - A static pointer hdac_acomp is used to help bind/unbind callbacks to get this component, because the sound card's private_data is used by the azx chip pointer, which is a legacy structure. It could be removed if private _data changes to some core structure which can be extended to find the bus. - snd_hdac_get_display_clk() is added to get the display core clock for HSW/BDW. - haswell_set_bclk() is moved to hda_intel.c because it needs to write the controller registers EM4/EM5, and only legacy HD-A needs it for HSW/BDW. - Move definition of HSW/BDW-specific registers EM4/EM5 to hda_register.h and rename them to HSW_EM4/HSW_EM5, because other HD-A controllers have different layout for the extended mode registers. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h new file mode 100644 index 0000000..adb5ba5 --- /dev/null +++ b/include/sound/hda_i915.h @@ -0,0 +1,36 @@ +/* + * HD-Audio helpers to sync with i915 driver + */ +#ifndef __SOUND_HDA_I915_H +#define __SOUND_HDA_I915_H + +#ifdef CONFIG_SND_HDA_I915 +int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); +int snd_hdac_display_power(struct hdac_bus *bus, bool enable); +int snd_hdac_get_display_clk(struct hdac_bus *bus); +int snd_hdac_i915_init(struct hdac_bus *bus); +int snd_hdac_i915_exit(struct hdac_bus *bus); +#else +static int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int snd_hdac_get_display_clk(struct hdac_bus *bus) +{ + return 0; +} +static inline int snd_hdac_i915_init(struct hdac_bus *bus) +{ + return -ENODEV; +} +static inline int snd_hdac_i915_exit(struct hdac_bus *bus) +{ + return 0; +} +#endif + +#endif /* __SOUND_HDA_I915_H */ diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 4f6d3fc..0c7536e 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -84,6 +84,10 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_SD_BDLPL 0x18 #define AZX_REG_SD_BDLPU 0x1c +/* Haswell/Broadwell display HD-A controller Extended Mode registers */ +#define AZX_REG_HSW_EM4 0x100c +#define AZX_REG_HSW_EM5 0x1010 + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b97c59e..64fff4d 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -11,6 +11,7 @@ #include <sound/core.h> #include <sound/memalloc.h> #include <sound/hda_verbs.h> +#include <drm/i915_component.h> /* codec node id */ typedef u16 hda_nid_t; @@ -285,6 +286,10 @@ struct hdac_bus { /* locks */ spinlock_t reg_lock; struct mutex cmd_mutex; + + /* i915 component interface */ + struct i915_audio_component *audio_component; + int i915_power_refcount; }; int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 7a17fca..ac5ffac 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -4,3 +4,9 @@ config SND_HDA_CORE config SND_HDA_DSP_LOADER bool + +config SND_HDA_I915 + bool + default y + depends on DRM_I915 + depends on SND_HDA_CORE diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 5b4bb47..55dd465 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -4,4 +4,7 @@ snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o \ snd-hda-core-objs += trace.o CFLAGS_trace.o := -I$(src) +# for sync with i915 gfx driver +snd-hda-core-$(CONFIG_SND_HDA_I915) += hdac_i915.o + obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c new file mode 100644 index 0000000..cb78c25 --- /dev/null +++ b/sound/hda/hdac_i915.c @@ -0,0 +1,193 @@ +/* + * hdac_i915.c - routines for sync between HD-A core and i915 display driver + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY + * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License + * for more details. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/component.h> +#include <drm/i915_component.h> +#include <sound/core.h> +#include <sound/hdaudio.h> +#include <sound/hda_i915.h> + +static struct i915_audio_component *hdac_acomp; + +int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + if (!acomp->ops->codec_wake_override) { + dev_warn(bus->dev, + "Invalid codec wake callback\n"); + return 0; + } + + dev_dbg(bus->dev, "%s codec wakeup\n", + enable ? "enable" : "disable"); + + acomp->ops->codec_wake_override(acomp->dev, enable); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); + +int snd_hdac_display_power(struct hdac_bus *bus, bool enable) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + dev_dbg(bus->dev, "display power %s\n", + enable ? "enable" : "disable"); + + if (enable) { + if (!bus->i915_power_refcount++) + acomp->ops->get_power(acomp->dev); + } else { + WARN_ON(!bus->i915_power_refcount); + if (!--bus->i915_power_refcount) + acomp->ops->put_power(acomp->dev); + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_display_power); + +int snd_hdac_get_display_clk(struct hdac_bus *bus) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + return acomp->ops->get_cdclk_freq(acomp->dev); +} +EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); + +static int hdac_component_master_bind(struct device *dev) +{ + struct i915_audio_component *acomp = hdac_acomp; + int ret; + + ret = component_bind_all(dev, acomp); + if (ret < 0) + return ret; + + if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && + acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { + ret = -EINVAL; + goto out_unbind; + } + + /* + * Atm, we don't support dynamic unbinding initiated by the child + * component, so pin its containing module until we unbind. + */ + if (!try_module_get(acomp->ops->owner)) { + ret = -ENODEV; + goto out_unbind; + } + + return 0; + +out_unbind: + component_unbind_all(dev, acomp); + + return ret; +} + +static void hdac_component_master_unbind(struct device *dev) +{ + struct i915_audio_component *acomp = hdac_acomp; + + module_put(acomp->ops->owner); + component_unbind_all(dev, acomp); + WARN_ON(acomp->ops || acomp->dev); +} + +static const struct component_master_ops hdac_component_master_ops = { + .bind = hdac_component_master_bind, + .unbind = hdac_component_master_unbind, +}; + +static int hdac_component_master_match(struct device *dev, void *data) +{ + /* i915 is the only supported component */ + return !strcmp(dev->driver->name, "i915"); +} + +int snd_hdac_i915_init(struct hdac_bus *bus) +{ + struct component_match *match = NULL; + struct device *dev = bus->dev; + struct i915_audio_component *acomp; + int ret; + + acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); + if (!acomp) + return -ENOMEM; + bus->audio_component = acomp; + hdac_acomp = acomp; + + component_match_add(dev, &match, hdac_component_master_match, bus); + ret = component_master_add_with_match(dev, &hdac_component_master_ops, + match); + if (ret < 0) + goto out_err; + + /* + * Atm, we don't support deferring the component binding, so make sure + * i915 is loaded and that the binding successfully completes. + */ + request_module("i915"); + + if (!acomp->ops) { + ret = -ENODEV; + goto out_master_del; + } + dev_dbg(dev, "bound to i915 component master\n"); + + return 0; +out_master_del: + component_master_del(dev, &hdac_component_master_ops); +out_err: + kfree(acomp); + bus->audio_component = NULL; + dev_err(dev, "failed to add i915 component master (%d)\n", ret); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_i915_init); + +int snd_hdac_i915_exit(struct hdac_bus *bus) +{ + struct device *dev = bus->dev; + struct i915_audio_component *acomp = bus->audio_component; + + WARN_ON(bus->i915_power_refcount); + if (bus->i915_power_refcount > 0 && acomp && acomp->ops) + acomp->ops->put_power(acomp->dev); + + component_master_del(dev, &hdac_component_master_ops); + + kfree(acomp); + bus->audio_component = NULL; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_i915_exit); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 117bf5c..98ced97 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -145,11 +145,6 @@ config SND_HDA_CODEC_HDMI comment "Set to Y if you want auto-loading the codec driver" depends on SND_HDA=y && SND_HDA_CODEC_HDMI=m -config SND_HDA_I915 - bool - default y - depends on DRM_I915 - config SND_HDA_CODEC_CIRRUS tristate "Build Cirrus Logic codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 9c259ce..90e69b2 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,7 +1,5 @@ snd-hda-intel-objs := hda_intel.o snd-hda-tegra-objs := hda_tegra.o -# for haswell power well -snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o snd-hda-codec-y := hda_bind.o hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o snd-hda-codec-y += hda_controller.o diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c deleted file mode 100644 index 8521702..0000000 --- a/sound/pci/hda/hda_i915.c +++ /dev/null @@ -1,227 +0,0 @@ -/* - * hda_i915.c - routines for Haswell HDA controller power well support - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the Free - * Software Foundation; either version 2 of the License, or (at your option) - * any later version. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY - * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License - * for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software Foundation, - * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/pci.h> -#include <linux/component.h> -#include <drm/i915_component.h> -#include <sound/core.h> -#include "hda_controller.h" -#include "hda_intel.h" - -/* Intel HSW/BDW display HDA controller Extended Mode registers. - * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display - * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled. - */ -#define AZX_REG_EM4 0x100c -#define AZX_REG_EM5 0x1010 - -int hda_set_codec_wakeup(struct hda_intel *hda, bool enable) -{ - struct i915_audio_component *acomp = &hda->audio_component; - - if (!acomp->ops) - return -ENODEV; - - if (!acomp->ops->codec_wake_override) { - dev_warn(&hda->chip.pci->dev, - "Invalid codec wake callback\n"); - return 0; - } - - dev_dbg(&hda->chip.pci->dev, "%s codec wakeup\n", - enable ? "enable" : "disable"); - - acomp->ops->codec_wake_override(acomp->dev, enable); - - return 0; -} - -int hda_display_power(struct hda_intel *hda, bool enable) -{ - struct i915_audio_component *acomp = &hda->audio_component; - - if (!acomp->ops) - return -ENODEV; - - dev_dbg(&hda->chip.pci->dev, "display power %s\n", - enable ? "enable" : "disable"); - - if (enable) { - if (!hda->i915_power_refcount++) - acomp->ops->get_power(acomp->dev); - } else { - WARN_ON(!hda->i915_power_refcount); - if (!--hda->i915_power_refcount) - acomp->ops->put_power(acomp->dev); - } - - return 0; -} - -void haswell_set_bclk(struct hda_intel *hda) -{ - int cdclk_freq; - unsigned int bclk_m, bclk_n; - struct i915_audio_component *acomp = &hda->audio_component; - struct pci_dev *pci = hda->chip.pci; - - /* Only Haswell/Broadwell need set BCLK */ - if (pci->device != 0x0a0c && pci->device != 0x0c0c - && pci->device != 0x0d0c && pci->device != 0x160c) - return; - - if (!acomp->ops) - return; - - cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev); - switch (cdclk_freq) { - case 337500: - bclk_m = 16; - bclk_n = 225; - break; - - case 450000: - default: /* default CDCLK 450MHz */ - bclk_m = 4; - bclk_n = 75; - break; - - case 540000: - bclk_m = 4; - bclk_n = 90; - break; - - case 675000: - bclk_m = 8; - bclk_n = 225; - break; - } - - azx_writew(&hda->chip, EM4, bclk_m); - azx_writew(&hda->chip, EM5, bclk_n); -} - -static int hda_component_master_bind(struct device *dev) -{ - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - struct i915_audio_component *acomp = &hda->audio_component; - int ret; - - ret = component_bind_all(dev, acomp); - if (ret < 0) - return ret; - - if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && - acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { - ret = -EINVAL; - goto out_unbind; - } - - /* - * Atm, we don't support dynamic unbinding initiated by the child - * component, so pin its containing module until we unbind. - */ - if (!try_module_get(acomp->ops->owner)) { - ret = -ENODEV; - goto out_unbind; - } - - return 0; - -out_unbind: - component_unbind_all(dev, acomp); - - return ret; -} - -static void hda_component_master_unbind(struct device *dev) -{ - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - struct i915_audio_component *acomp = &hda->audio_component; - - module_put(acomp->ops->owner); - component_unbind_all(dev, acomp); - WARN_ON(acomp->ops || acomp->dev); -} - -static const struct component_master_ops hda_component_master_ops = { - .bind = hda_component_master_bind, - .unbind = hda_component_master_unbind, -}; - -static int hda_component_master_match(struct device *dev, void *data) -{ - /* i915 is the only supported component */ - return !strcmp(dev->driver->name, "i915"); -} - -int hda_i915_init(struct hda_intel *hda) -{ - struct component_match *match = NULL; - struct device *dev = &hda->chip.pci->dev; - struct i915_audio_component *acomp = &hda->audio_component; - int ret; - - component_match_add(dev, &match, hda_component_master_match, hda); - ret = component_master_add_with_match(dev, &hda_component_master_ops, - match); - if (ret < 0) - goto out_err; - - /* - * Atm, we don't support deferring the component binding, so make sure - * i915 is loaded and that the binding successfully completes. - */ - request_module("i915"); - - if (!acomp->ops) { - ret = -ENODEV; - goto out_master_del; - } - - dev_dbg(dev, "bound to i915 component master\n"); - - return 0; -out_master_del: - component_master_del(dev, &hda_component_master_ops); -out_err: - dev_err(dev, "failed to add i915 component master (%d)\n", ret); - - return ret; -} - -int hda_i915_exit(struct hda_intel *hda) -{ - struct device *dev = &hda->chip.pci->dev; - struct i915_audio_component *acomp = &hda->audio_component; - - WARN_ON(hda->i915_power_refcount); - if (hda->i915_power_refcount > 0 && acomp->ops) - acomp->ops->put_power(acomp->dev); - - component_master_del(dev, &hda_component_master_ops); - - return 0; -} diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5c84d40..391e4f8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -57,6 +57,8 @@ #endif #include <sound/core.h> #include <sound/initval.h> +#include <sound/hdaudio.h> +#include <sound/hda_i915.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -496,13 +498,13 @@ static void azx_init_pci(struct azx *chip) static void hda_intel_init_chip(struct azx *chip, bool full_reset) { - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - hda_set_codec_wakeup(hda, true); + snd_hdac_set_codec_wakeup(bus, true); azx_init_chip(chip, full_reset); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - hda_set_codec_wakeup(hda, false); + snd_hdac_set_codec_wakeup(bus, false); } /* calculate runtime delay from LPIB */ @@ -560,9 +562,9 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) /* Enable/disable i915 display power for the link */ static int azx_intel_link_power(struct azx *chip, bool enable) { - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); - return hda_display_power(hda, enable); + return snd_hdac_display_power(bus, enable); } /* @@ -800,6 +802,50 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ +/* Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK + * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) + * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: + * BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled and need to + * be restored to avoid abnormal playback speed. + */ +static void haswell_set_bclk(struct hda_intel *hda) +{ + struct azx *chip = &hda->chip; + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!hda->need_i915_power) + return; + + cdclk_freq = snd_hdac_get_display_clk(azx_bus(chip)); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } + + azx_writew(chip, HSW_EM4, bclk_m); + azx_writew(chip, HSW_EM5, bclk_n); +} + #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -833,7 +879,7 @@ static int azx_suspend(struct device *dev) pci_disable_msi(chip->pci); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) - hda_display_power(hda, false); + snd_hdac_display_power(bus, false); trace_azx_suspend(chip); return 0; @@ -856,7 +902,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { - hda_display_power(hda, true); + snd_hdac_display_power(azx_bus(chip), true); haswell_set_bclk(hda); } if (chip->msi) @@ -902,7 +948,7 @@ static int azx_runtime_suspend(struct device *dev) azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) - hda_display_power(hda, false); + snd_hdac_display_power(azx_bus(chip), false); trace_azx_runtime_suspend(chip); return 0; @@ -913,6 +959,7 @@ static int azx_runtime_resume(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; struct hda_intel *hda; + struct hdac_bus *bus; struct hda_codec *codec; int status; @@ -929,11 +976,12 @@ static int azx_runtime_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { - hda_display_power(hda, true); + bus = azx_bus(chip); + snd_hdac_display_power(bus, true); haswell_set_bclk(hda); /* toggle codec wakeup bit for STATESTS read */ - hda_set_codec_wakeup(hda, true); - hda_set_codec_wakeup(hda, false); + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); } /* Read STATESTS before controller reset */ @@ -1152,10 +1200,11 @@ static int azx_free(struct azx *chip) #ifdef CONFIG_SND_HDA_PATCH_LOADER release_firmware(chip->fw); #endif + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { if (hda->need_i915_power) - hda_display_power(hda, false); - hda_i915_exit(hda); + snd_hdac_display_power(bus, false); + snd_hdac_i915_exit(bus); } kfree(hda); @@ -1914,6 +1963,7 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { static int azx_probe_continue(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; int dev = chip->dev_index; int err; @@ -1930,11 +1980,11 @@ static int azx_probe_continue(struct azx *chip) if (pci->device != 0x0f04 && pci->device != 0x2284) hda->need_i915_power = 1; - err = hda_i915_init(hda); + err = snd_hdac_i915_init(bus); if (err < 0) goto i915_power_fail; - err = hda_display_power(hda, true); + err = snd_hdac_display_power(bus, true); if (err < 0) { dev_err(chip->card->dev, "Cannot turn on display power on i915\n"); @@ -1986,7 +2036,7 @@ static int azx_probe_continue(struct azx *chip) out_free: if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && !hda->need_i915_power) - hda_display_power(hda, false); + snd_hdac_display_power(bus, false); i915_power_fail: if (err < 0) diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 7fd3254..354f0bb 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -16,7 +16,6 @@ #ifndef __SOUND_HDA_INTEL_H #define __SOUND_HDA_INTEL_H -#include <drm/i915_component.h> #include "hda_controller.h" struct hda_intel { @@ -44,36 +43,7 @@ struct hda_intel { /* secondary power domain for hdmi audio under vga device */ struct dev_pm_domain hdmi_pm_domain; - /* i915 component interface */ bool need_i915_power:1; /* the hda controller needs i915 power */ - struct i915_audio_component audio_component; - int i915_power_refcount; }; -#ifdef CONFIG_SND_HDA_I915 -int hda_set_codec_wakeup(struct hda_intel *hda, bool enable); -int hda_display_power(struct hda_intel *hda, bool enable); -void haswell_set_bclk(struct hda_intel *hda); -int hda_i915_init(struct hda_intel *hda); -int hda_i915_exit(struct hda_intel *hda); -#else -static inline int hda_set_codec_wakeup(struct hda_intel *hda, bool enable) -{ - return 0; -} -static inline int hda_display_power(struct hda_intel *hda, bool enable) -{ - return 0; -} -static inline void haswell_set_bclk(struct hda_intel *hda) { return; } -static inline int hda_i915_init(struct hda_intel *hda) -{ - return 0; -} -static inline int hda_i915_exit(struct hda_intel *hda) -{ - return 0; -} -#endif - #endif -- 1.9.1

2 1

[alsa-devel] Focusrite Forte
by e-rouge 19 May '15

19 May '15

Hello, I just bought a Focusrite Forte, and yeah I've fooled by the class-compliant thingy and the touch controls. For some reasons the card controls require a deamon that only runs on window and mac. Well, OK, I'm a programmer, I can fix that. There must be a way to access all the parameters (namely gain and phantom switches). Can you provide me some basic commands to know what this card has under the hood ? What tools do you use to do that ? Thanks a lot in advance T. P.S. : well if someone has a solution out-of-the-box, I can take it too :)

1 0

Re: [alsa-devel] linux-next: Tree for May 19 (sound/soc/atmel/)
by Randy Dunlap 19 May '15

19 May '15

On 05/19/15 00:30, Stephen Rothwell wrote: > Hi all, > > Changes since 20150518: > on x86_64: sound/built-in.o: In function `atmel_ssc_set_audio': (.text+0x87d90): undefined reference to `atmel_pcm_pdc_platform_register' sound/built-in.o: In function `atmel_ssc_put_audio': (.text+0x8879a): undefined reference to `atmel_pcm_pdc_platform_unregister' Full randconfig file is attached. -- ~Randy

1 0

Re: [alsa-devel] USB headset cannot get freq
by Greg KH 19 May '15

19 May '15

On Sat, May 16, 2015 at 04:25:23PM +0200, Heinz Diehl wrote: > Hi, > > this is the Jabra Evolve 40 MS headset: > Bus 001 Device 008: ID 0b0e:0305 GN Netcom > > When plugged in, dmesg says: > > [ 247.215998] usb 1-5: new full-speed USB device number 8 using xhci_hcd > [ 247.384000] usb 1-5: New USB device found, idVendor=0b0e, idProduct=0305 > [ 247.384008] usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > [ 247.384012] usb 1-5: Product: Jabra EVOLVE LINK MS > [ 247.384016] usb 1-5: Manufacturer: GN Netcom A/S > [ 247.384019] usb 1-5: SerialNumber: 0001A3E08CFC09 > [ 247.386542] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.388490] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.406532] input: GN Netcom A/S Jabra EVOLVE LINK MS as /devices/pci0000:00/0000:00:14.0/usb1/1-5/1-5:1.3/0003:0B0E:0305.0009/input/input12 > [ 247.457203] hid-generic 0003:0B0E:0305.0009: input,hiddev0,hidraw1: USB HID v1.00 Device [GN Netcom A/S Jabra EVOLVE LINK MS] on usb-0000:00:14.0-5/input3 > [ 247.467683] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.467912] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.473505] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.473718] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.475488] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.475721] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.481899] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.482097] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.483351] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.483566] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.487396] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.487586] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.494730] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.495018] usb 1-5: 1:1: cannot get freq at ep 0x83 > > The micro/playback devices show up, but there's no sound and the mic is also > dead. > > I've spent the last two days reading on the net, tried a lot of different > options, but whithout any success. In a lot of similar bug reports, people > point out that this is a bug in the Linux kernel USB subsystem.. > > What can I do to debug this further, in order to get a working headset? The sound developers should be able to help you (added to the cc:) greg k-h

3 2

[alsa-devel] [PATCH v3 00/17] new LoCoMo driver set
by Dmitry Eremin-Solenikov 19 May '15

19 May '15

Sharp Zaurus SL-5500 and SL-5600 use special companion Gate Array. With this patchset I tried to modernise and restructure the LoCoMo drivers. * Use platform bus and mfd core code * Use GPIO API instead of custom locomo GPIO code * Use irqdomains to manage IRQs * Split mixed locomolcd driver to lcd and backlight parts * Implement IRQ handling in GPIO driver * Implement SPI driver used as a host for mmc_spi device to control SD cards Changes since V2: * Implemented most of review findings (lots of small changes) * Added comments where the review feedback was ignored The following changes since commit b787f68c36d49bb1d9236f403813641efa74a031: Linux 4.1-rc1 (2015-04-26 17:59:10 -0700) are available in the git repository at: git://git.infradead.org/users/dbaryshkov/zaurus.git locomo-v3 for you to fetch changes up to c7405f110836863287c505650ac2133fecabea7b: ARM: drop old LoCoMo driver (2015-05-17 19:23:20 +0300) ---------------------------------------------------------------- Dmitry Eremin-Solenikov (17): mfd: add new driver for Sharp LoCoMo leds: port locomo leds driver to new locomo core input: convert LoCoMo keyboard driver to use new locomo core input: make LoCoMo keyboard driver support both poodle and collie video: backlight: add new locomo backlight driver video: lcd: add LoCoMo LCD driver gpio: port LoCoMo gpio support from old driver gpio: locomo: implement per-pin irq handling spi: add locomo SPI driver i2c: add locomo i2c driver ARM: sa1100: make collie use new locomo drivers ARM: sa1100: don't preallocate IRQ space for locomo ASoC: pxa: poodle: make use of new locomo GPIO interface ARM: pxa: poodle: use new LoCoMo driver ARM: pxa: poodle: don't preallocate IRQ space for locomo video: backlight: drop old locomo bl/lcd driver ARM: drop old LoCoMo driver arch/arm/common/Kconfig | 3 - arch/arm/common/Makefile | 1 - arch/arm/common/locomo.c | 914 ----------------------------- arch/arm/include/asm/hardware/locomo.h | 221 ------- arch/arm/mach-pxa/Kconfig | 1 - arch/arm/mach-pxa/include/mach/poodle.h | 8 +- arch/arm/mach-pxa/poodle.c | 58 +- arch/arm/mach-sa1100/Kconfig | 1 - arch/arm/mach-sa1100/collie.c | 213 +++++-- arch/arm/mach-sa1100/include/mach/collie.h | 16 +- arch/arm/mach-sa1100/include/mach/irqs.h | 19 +- drivers/gpio/Kconfig | 13 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-locomo.c | 285 +++++++++ drivers/i2c/busses/Kconfig | 12 + drivers/i2c/busses/Makefile | 1 + drivers/i2c/busses/i2c-locomo.c | 135 +++++ drivers/input/keyboard/Kconfig | 2 +- drivers/input/keyboard/locomokbd.c | 264 ++++----- drivers/leds/Kconfig | 2 +- drivers/leds/leds-locomo.c | 117 ++-- drivers/mfd/Kconfig | 10 + drivers/mfd/Makefile | 1 + drivers/mfd/locomo.c | 346 +++++++++++ drivers/spi/Kconfig | 10 + drivers/spi/Makefile | 1 + drivers/spi/spi-locomo.c | 332 +++++++++++ drivers/video/backlight/Kconfig | 16 +- drivers/video/backlight/Makefile | 3 +- drivers/video/backlight/locomo_bl.c | 153 +++++ drivers/video/backlight/locomo_lcd.c | 285 +++++++++ drivers/video/backlight/locomolcd.c | 255 -------- include/linux/mfd/locomo.h | 167 ++++++ sound/soc/pxa/poodle.c | 52 +- 34 files changed, 2216 insertions(+), 1702 deletions(-) delete mode 100644 arch/arm/common/locomo.c delete mode 100644 arch/arm/include/asm/hardware/locomo.h create mode 100644 drivers/gpio/gpio-locomo.c create mode 100644 drivers/i2c/busses/i2c-locomo.c create mode 100644 drivers/mfd/locomo.c create mode 100644 drivers/spi/spi-locomo.c create mode 100644 drivers/video/backlight/locomo_bl.c create mode 100644 drivers/video/backlight/locomo_lcd.c delete mode 100644 drivers/video/backlight/locomolcd.c create mode 100644 include/linux/mfd/locomo.h -- With best wishes Dmitry

5 26

[alsa-devel] [PATCH] ASoC: rt5645: Rename HP control to Headphone
by Nicolas Boichat 19 May '15

19 May '15

Use the standard name "Headphone" instead of "HP" for better userspace integration. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- 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 be4d741..bc925f0 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -464,9 +464,9 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Channel Switch", RT5645_HP_VOL, + SOC_DOUBLE("Headphone Channel Switch", RT5645_HP_VOL, RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), - SOC_DOUBLE_TLV("HP Playback Volume", RT5645_HP_VOL, + SOC_DOUBLE_TLV("Headphone Playback Volume", RT5645_HP_VOL, RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), /* OUTPUT Control */ -- 2.2.0.rc0.207.ga3a616c

4 4

[alsa-devel] [PATCH] ASoC: rt5645: fix kernel hang when call rt5645_set_jack_detect()
by John Lin 19 May '15

19 May '15

rt5645_set_jack_detect() is usually called from snd_soc_dai_link.init() and it calls snd_soc_jack_report() from rt5645_irq_detection() if jack is inserted. snd_soc_jack_report() results in kernel hang if it is called from a context which cannot sleep. This patch makes sure snd_soc_jack_report() is called from workqueue. It can fix the kernel hang issue. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 14b12c5..aaede45 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2876,6 +2876,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } static int rt5645_irq_detection(struct rt5645_priv *rt5645); +static irqreturn_t rt5645_irq(int irq, void *data); int rt5645_set_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, @@ -2895,7 +2896,7 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); } - rt5645_irq_detection(rt5645); + rt5645_irq(0, rt5645); return 0; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix NULL dereference from CA0132 DSP loader
by Takashi Iwai 19 May '15

19 May '15

The CA0132 DSP loader leads to NULL deference since the recent transition to HDA core code, as it unconditionally accesses hdac_stream->substream->runtime. For DSP loading, the substream shouldn't be assigned. This patch addresses the NULL dereference above in addition to assure the substream is cleared while DSP loading. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98151 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_stream.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 52a894fe478f..4c15d0accc9e 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -139,9 +139,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_reset); int snd_hdac_stream_setup(struct hdac_stream *azx_dev) { struct hdac_bus *bus = azx_dev->bus; - struct snd_pcm_runtime *runtime = azx_dev->substream->runtime; + struct snd_pcm_runtime *runtime; unsigned int val; + if (azx_dev->substream) + runtime = azx_dev->substream->runtime; + else + runtime = NULL; /* make sure the run bit is zero for SD */ snd_hdac_stream_clear(azx_dev); /* program the stream_tag */ @@ -189,14 +193,15 @@ int snd_hdac_stream_setup(struct hdac_stream *azx_dev) * we ignore it; currently set the threshold statically to * 64 frames */ - if (runtime->period_size > 64) + if (runtime && runtime->period_size > 64) azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64); else azx_dev->delay_negative_threshold = 0; /* wallclk has 24Mhz clock source */ - azx_dev->period_wallclk = (((runtime->period_size * 24000) / + if (runtime) + azx_dev->period_wallclk = (((runtime->period_size * 24000) / runtime->rate) * 1000); return 0; @@ -611,6 +616,7 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format, if (err < 0) goto err_alloc; + azx_dev->substream = NULL; azx_dev->bufsize = byte_size; azx_dev->period_bytes = byte_size; azx_dev->format_val = format; -- 2.4.1

1 0

[alsa-devel] [PATCH 0/8] ASoC: Intel: refactor common SST IPC handling
by Vinod Koul 19 May '15

19 May '15

This series refactors the common SST IPC code to remove some hardcoded assumption which are no longer valid for future platforms like Skylake, These are made configurable and right configuration applied for respective drivers. The changes are mailbox sizes and dsp busy checks. Subhransu S. Prusty (8): ASoC: Intel: Create an ops to check for DSP busy ASoC: Intel: Move the busy check to ops for Baytrail ASoC: Intel: Move the busy check to ops for HSW ASoC: Intel: Remove the direct register reference from common ipc ASoC: Intel: Allow to configure max size for mailbox data ASoC: Intel: Initialize max mailbox size for baytrail ASoC: Intel: Initialize max mailbox size for haswell ASoC: Intel: Allocate for the mailbox with max size sound/soc/intel/baytrail/sst-baytrail-ipc.c | 11 +++++++++++ sound/soc/intel/common/sst-ipc.c | 20 +++++++++++++++++--- sound/soc/intel/common/sst-ipc.h | 7 +++++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 12 ++++++++++++ 4 files changed, 45 insertions(+), 5 deletions(-) -- 1.9.1

1 9

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for MS LifeCam Studio
by Takashi Iwai 19 May '15

19 May '15

Microsoft LifeCam Studio (045e:0772) needs a similar quirk for suppressing the wrong sample rate inquiry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7c5a70139278..46facfc9aec1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1117,6 +1117,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) switch (chip->usb_id) { case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema */ case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ + case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ return true; } -- 2.4.1

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Add headset-mode* model options for ALC269 & co
by Takashi Iwai 19 May '15

19 May '15

Modern machines tend to have only one headset jack nowadays, and they often need these quirks. Let's allow them applicable via model option for ease of debugging. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio-Models.txt | 2 ++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 4 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 5a3163cac6c3..d92b85d38adc 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -29,6 +29,8 @@ ALC269/270/275/276/28x/29x alc271-dmic Enable ALC271X digital mic workaround inv-dmic Inverted internal mic workaround headset-mic Indicates a combined headset (headphone+mic) jack + headset-mode More comprehensive headset support for ALC269 & co + headset-mode-no-hp-mic Headset mode support without headphone mic lenovo-dock Enables docking station I/O for some Lenovos dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6e1c8c34e841..4d11fbfca5e2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5242,6 +5242,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC269_FIXUP_HEADSET_MIC, .name = "headset-mic"}, + {.id = ALC269_FIXUP_HEADSET_MODE, .name = "headset-mode"}, + {.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, -- 2.4.1

1 1

[alsa-devel] [RFC PATCH] ALSA: pcm: Modify double acknowledged interrupts check condition
by Koro Chen 19 May '15

19 May '15

Currently in snd_pcm_update_hw_ptr0 during interrupt, we consider there were double acknowledged interrupts when: 1. HW reported pointer is smaller than expected, and 2. Time from last update time (hdelta) is over half a buffer time. However, when HW reported pointer is only a few bytes smaller than expected, and when hdelta is just a little larger than half a buffer time (e.g. ping-pong buffer), it wrongly treats this IRQ as double acknowledged. The condition #2 uses jiffies, but jiffies is not high resolution since it is integer. We should consider jiffies inaccuracy. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ac6b33f..7d45645 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -339,7 +339,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (delta > new_hw_ptr) { /* check for double acknowledged interrupts */ hdelta = curr_jiffies - runtime->hw_ptr_jiffies; - if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { + if (hdelta > runtime->hw_ptr_buffer_jiffies/2 + 1) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) { hw_base = 0; -- 1.8.1.1.dirty

2 1

[alsa-devel] Intel HDA / ca0132: support for Alienware 15 Creative Sound Core3D-EX
by Gabriele Martino 19 May '15

19 May '15

Hi, the new Alienware 15 laptop (2015, not the M15x) has a Creative Sound Core3D-EX Audio which seems to work with ca0132. I'm using a 4.0.0 kernel with gentoo patchset, but with the default configuration: - the line out/headphone jack doesn't work, it only outputs some white noise - alsamixer shows only "Master" and "PCM" channel, no "Front", "Line Out" or similar - alsamixer has a toggle switch for the "HP/Speaker" but no headphone/line out switch - the jack detection is enabled on the "line in" jack instead of the "line out" (if I plug something on the "line in", the speakers are muted) - if I play too much with these switches, the speakers stop working until suspend/reboot To fix the first issue I run "hdajacksensetest" to get the headphone jack pin: hdajacksensetest -c 1 -a Pin 0x0b (Internal Speaker): present = Yes Pin 0x0c (Not connected): present = No Pin 0x0d (Not connected): present = No Pin 0x0e (Not connected): present = No Pin 0x0f (Not connected): present = Yes Pin 0x10 (Not connected): present = No Pin 0x11 (Black Line In, Left side): present = No Pin 0x12 (Internal Mic, Mobile-In): present = No Pin 0x13 (Not connected): present = No Pin 0x18 (Not connected): present = No and replaced the pin number: --- /usr/src/linux-4.0.0-gentoo/sound/pci/hda/patch_ca0132.c.old 2015-04-13 00:12:50.000000000 +0200 +++ /usr/src/linux-4.0.0-gentoo/sound/pci/hda/patch_ca0132.c 2015-04-25 01:43:09.607531550 +0200 @@ -4625,7 +4625,7 @@ spec->num_outputs = 2; spec->out_pins[0] = 0x0b; /* speaker out */ - spec->out_pins[1] = 0x10; /* headphone out */ + spec->out_pins[1] = 0x0f; /* default: 0x10 - headphone out -0x0f on Alienware 15 2015 */ spec->shared_out_nid = 0x2; spec->num_inputs = 3; Now the "HP/Speaker Auto Detect" is ignored on both jacks, but I can switch between headphones and speakers manually with the "HP/Speaker" item. This is a dirty hack (and probably won't work on different cards), is there a better way to fix this and get at least the jack detection working? Regards, Gabriele

2 20

[alsa-devel] [PATCH 0/4] firewire-lib: purge restriction of synchronization for non-blocking mode
by Takashi Sakamoto 19 May '15

19 May '15

This patchset allows non-blocking streams to use timestamp synchronization. In full duplex synchronization mode, current implementation pass the value of 'syt' field of CIP header in incoming packets to outgoing packet processing, while the number of data blocks is not passed. Therefore, this mode is allowed just to blocking streams, because of a lack of validator for the number which actual devices transfers. As long as the number is suspicious, it's not reused to outgoing packet processing. To allow the synchronization mode to non-blocking streams, this patchset: * adds a validator for the number of data blocks in incoming packets * adds a flag still to support illegal models (i.e. Behringer FCA202) * pass the evaluated number from incoming packet processing to outgoing * remove the restriction Takashi Sakamoto (4): ALSA: firewire-lib: add buffer-over-run protection at receiving more data blocks than expected ALSA: firewire-lib: simplify function to calculate the number of data blocks ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets ALSA: firewire-lib: remove restriction for non-blocking mode sound/firewire/amdtp.c | 108 ++++++++++++++++++++++++-------------- sound/firewire/amdtp.h | 4 ++ sound/firewire/oxfw/oxfw-stream.c | 10 +++- 3 files changed, 81 insertions(+), 41 deletions(-) -- 2.1.4

2 8

[alsa-devel] [PATCH RESEND] sound: ppc: keywest: drop using attach adapter
by Wolfram Sang 18 May '15

18 May '15

As we now have deferred probing, we can use a custom mechanism and finally get rid of this legacy interface from the i2c core. Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> Tested-by: Dan DeVoto <dand1972(a)yahoo.com> Tested-by: Mark Elliott <txlitebeer(a)gmail.com> --- Changes since last time: Added two test tags from the Debian PPC list. I'd love to have that in 4.2, so I can get rid of that legacy interface. I can take it via I2C but ALSA looks like the proper path to me. sound/ppc/keywest.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 0d1c27e911b8dd..d7627bae08362e 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -52,7 +52,7 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) return -EINVAL; if (strncmp(adapter->name, "mac-io", 6)) - return 0; /* ignored */ + return -EINVAL; /* ignored */ memset(&info, 0, sizeof(struct i2c_board_info)); strlcpy(info.type, "keywest", I2C_NAME_SIZE); @@ -100,7 +100,6 @@ static struct i2c_driver keywest_driver = { .driver = { .name = "PMac Keywest Audio", }, - .attach_adapter = keywest_attach_adapter, .probe = keywest_probe, .remove = keywest_remove, .id_table = keywest_i2c_id, @@ -132,16 +131,32 @@ int snd_pmac_tumbler_post_init(void) /* exported */ int snd_pmac_keywest_init(struct pmac_keywest *i2c) { - int err; + struct i2c_adapter *adap; + int err, i = 0; if (keywest_ctx) return -EBUSY; + adap = i2c_get_adapter(0); + if (!adap) + return -EPROBE_DEFER; + keywest_ctx = i2c; if ((err = i2c_add_driver(&keywest_driver))) { snd_printk(KERN_ERR "cannot register keywest i2c driver\n"); + i2c_put_adapter(adap); return err; } - return 0; + + /* We assume Macs have consecutive I2C bus numbers starting at 0 */ + while (adap) { + err = keywest_attach_adapter(adap); + if (!err) + return 0; + i2c_put_adapter(adap); + adap = i2c_get_adapter(++i); + } + + return -ENODEV; } -- 2.1.4

2 1

[alsa-devel] Left/Right Mic mute independently
by Arijit De 18 May '15

18 May '15

Hi, I want to do Left/Right Mic mute independently on a Audio capture device(it has two audio channels), but I dont see that kind of control is provided in alsamixer(GUI) for capture devices. Is Left/Right Mic mute is supported in alsa mixer ? Thanks Arijit

2 1

[alsa-devel] [PATCH - alsa-plugins/pulse 1/1] Add 24 bit data formats to Pulse plugin.
by alex.d.wiggins＠gmail.com 18 May '15

18 May '15

From: Alex Wiggins <alex.d.wiggins(a)gmail.com> Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 0d3704b..5cb3452 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -994,7 +994,11 @@ static int pulse_hw_constraint(snd_pcm_pulse_t * pcm) SND_PCM_FORMAT_FLOAT_LE, SND_PCM_FORMAT_FLOAT_BE, SND_PCM_FORMAT_S32_LE, - SND_PCM_FORMAT_S32_BE + SND_PCM_FORMAT_S32_BE, + SND_PCM_FORMAT_S24_3LE, + SND_PCM_FORMAT_S24_3BE, + SND_PCM_FORMAT_S24_LE, + SND_PCM_FORMAT_S24_BE }; int err; -- 2.3.6

2 1

[alsa-devel] [PATCH] conf/ucm: broadwell-rt286: change to set capture volume and capture device
by Jie Yang 18 May '15

18 May '15

Set 'Mic Capture Volume' in capture device EnableSequence, to fix capture no volume by default issue. Also add JackHWMute Value item to mute onboard dmic while headset mic is plugged in. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- src/conf/ucm/broadwell-rt286/HiFi | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/src/conf/ucm/broadwell-rt286/HiFi b/src/conf/ucm/broadwell-rt286/HiFi index 58983ea..c09a01c 100644 --- a/src/conf/ucm/broadwell-rt286/HiFi +++ b/src/conf/ucm/broadwell-rt286/HiFi @@ -86,6 +86,7 @@ SectionDevice."Handset" { EnableSequence [ cdev "hw:broadwellrt286" + cset "name='Mic Capture Volume' 28" cset "name='ADC 0 Mux' 0" cset "name='ADC0 Capture Switch' on" cset "name='ADC0 Capture Volume' 127" @@ -99,6 +100,9 @@ SectionDevice."Handset" { Value { CaptureChannels "2" + JackDev "rt286-jack" + JackControl "Mic Jack" + JackHWMute "Mainmic" } } @@ -112,6 +116,7 @@ SectionDevice."Mainmic" { EnableSequence [ cdev "hw:broadwellrt286" + cset "name='Mic Capture Volume' 30" cset "name='ADC 0 Mux' 2" cset "name='ADC0 Capture Switch' on" cset "name='ADC0 Capture Volume' 127" -- 1.9.1

3 2

[alsa-devel] [PATCH] ucm: allow multiple devices in JackHWMute
by Tanu Kaskinen 18 May '15

18 May '15

One jack may mute multiple devices, so let's make JackHWMute a list of device names instead of just a single device name. Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Acked-by: Mark Brown <broonie(a)kernel.org> --- include/use-case.h | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) Changes in v2: - Fixed the typo that Arun pointed out. - Send to alsa-devel (v1 was accidentally not sent to the list). diff --git a/include/use-case.h b/include/use-case.h index e3308b1..c7789c0 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -311,14 +311,15 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * applications are likely to support only one or the other. * * If **JackHWMute** is set, it indicates that when the jack is plugged - * in, the hardware automatically mutes some other device. The - * JackHWMute value is the name of the muted device. Note that - * JackHWMute should be used only when the hardware enforces the - * automatic muting. If the hardware doesn't enforce any muting, it may - * still be tempting to set JackHWMute to trick upper software layers to - * e.g. automatically mute speakers when headphones are plugged in, but - * that's application policy configuration that doesn't belong to UCM - * configuration files. + * in, the hardware automatically mutes some other device(s). The + * JackHWMute value is a space-separated list of device names (this + * isn't compatible with device names with spaces in them, so don't use + * such device names!). Note that JackHWMute should be used only when + * the hardware enforces the automatic muting. If the hardware doesn't + * enforce any muting, it may still be tempting to set JackHWMute to + * trick upper software layers to e.g. automatically mute speakers when + * headphones are plugged in, but that's application policy + * configuration that doesn't belong to UCM configuration files. */ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, const char *identifier, -- 1.9.3

2 1

[alsa-devel] [PATCH] sound: oss: fix build warning
by Sudip Mukherjee 18 May '15

18 May '15

while building with allyesconfig it was giving a build warning about unused variable. declare the variable only if the driver is built as a module. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- build tested with allyesconfig and allmodconfig sound/oss/ad1848.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index ec1ee07..10c8de1 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -2860,6 +2860,7 @@ static struct { {NULL} }; +#ifdef MODULE static struct isapnp_device_id id_table[] = { { ISAPNP_VENDOR('C','M','I'), ISAPNP_DEVICE(0x0001), ISAPNP_VENDOR('@','@','@'), ISAPNP_FUNCTION(0x0001), 0 }, @@ -2877,6 +2878,7 @@ static struct isapnp_device_id id_table[] = { }; MODULE_DEVICE_TABLE(isapnp, id_table); +#endif static struct pnp_dev *activate_dev(char *devname, char *resname, struct pnp_dev *dev) { -- 1.8.1.2

2 1

[alsa-devel] [RFC PATCH] ALSA: hda - Move hda_i915.c from sound/pci/hda to sound/hda
by mengdong.lin＠intel.com 18 May '15

18 May '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The file is moved to hda core, so can be used by both legacy HDA driver and new Skylake audio driver. - The i915 audio component is moved to core bus and dynamically allocated. - A static pointer hda_acomp is used to help bind/unbind callbacks to get this component, because the sound card's private_data is used by the azx chip pointer, which is a legacy structure. It could be removed if private _data changes to some core structure which can be extended to find the bus. - hda_get_display_clk() is added to get the display core clock for HSW/BDW. - haswell_set_bclk() is moved to hda_intel.c because it needs to write the controller registers EM4/EM5. - Move definition of HSW/BDW-specific registers EM4/EM5 to hda_register.h and rename them to HSW_EM4/HSW_EM5, because other HD-A controllers have different layout for the extended mode registers. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h new file mode 100644 index 0000000..440e4c0 --- /dev/null +++ b/include/sound/hda_i915.h @@ -0,0 +1,36 @@ +/* + * HD-Audio helpers to sync with i915 driver + */ +#ifndef __SOUND_HDA_I915_H +#define __SOUND_HDA_I915_H + +#ifdef CONFIG_SND_HDA_I915 +int hda_set_codec_wakeup(struct hdac_bus *bus, bool enable); +int hda_display_power(struct hdac_bus *bus, bool enable); +int hda_get_display_clk(struct hdac_bus *bus); +int hda_i915_init(struct hdac_bus *bus); +int hda_i915_exit(struct hdac_bus *bus); +#else +static int hda_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int hda_display_power(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int hda_get_display_clk(struct hdac_bus *bus) +{ + return 0; +} +static inline int hda_i915_init(struct hdac_bus *bus) +{ + return -ENODEV; +} +static inline int hda_i915_exit(struct hdac_bus *bus) +{ + return 0; +} +#endif + +#endif /* __SOUND_HDA_I915_H */ diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 4f6d3fc..0c7536e 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -84,6 +84,10 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_SD_BDLPL 0x18 #define AZX_REG_SD_BDLPU 0x1c +/* Haswell/Broadwell display HD-A controller Extended Mode registers */ +#define AZX_REG_HSW_EM4 0x100c +#define AZX_REG_HSW_EM5 0x1010 + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b97c59e..64fff4d 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -11,6 +11,7 @@ #include <sound/core.h> #include <sound/memalloc.h> #include <sound/hda_verbs.h> +#include <drm/i915_component.h> /* codec node id */ typedef u16 hda_nid_t; @@ -285,6 +286,10 @@ struct hdac_bus { /* locks */ spinlock_t reg_lock; struct mutex cmd_mutex; + + /* i915 component interface */ + struct i915_audio_component *audio_component; + int i915_power_refcount; }; int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 7a17fca..ac5ffac 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -4,3 +4,9 @@ config SND_HDA_CORE config SND_HDA_DSP_LOADER bool + +config SND_HDA_I915 + bool + default y + depends on DRM_I915 + depends on SND_HDA_CORE diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 5b4bb47..1ce78c6 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -4,4 +4,7 @@ snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o \ snd-hda-core-objs += trace.o CFLAGS_trace.o := -I$(src) +# for sync with i915 gfx driver +snd-hda-core-$(CONFIG_SND_HDA_I915) += hda_i915.o + obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/hda_i915.c b/sound/hda/hda_i915.c new file mode 100644 index 0000000..5c87938 --- /dev/null +++ b/sound/hda/hda_i915.c @@ -0,0 +1,193 @@ +/* + * hda_i915.c - routines for Haswell HDA controller power well support + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY + * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License + * for more details. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/component.h> +#include <drm/i915_component.h> +#include <sound/core.h> +#include <sound/hdaudio.h> +#include <sound/hda_i915.h> + +static struct i915_audio_component *hda_acomp; + +int hda_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + if (!acomp->ops->codec_wake_override) { + dev_warn(bus->dev, + "Invalid codec wake callback\n"); + return 0; + } + + dev_dbg(bus->dev, "%s codec wakeup\n", + enable ? "enable" : "disable"); + + acomp->ops->codec_wake_override(acomp->dev, enable); + + return 0; +} +EXPORT_SYMBOL_GPL(hda_set_codec_wakeup); + +int hda_display_power(struct hdac_bus *bus, bool enable) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + dev_dbg(bus->dev, "display power %s\n", + enable ? "enable" : "disable"); + + if (enable) { + if (!bus->i915_power_refcount++) + acomp->ops->get_power(acomp->dev); + } else { + WARN_ON(!bus->i915_power_refcount); + if (!--bus->i915_power_refcount) + acomp->ops->put_power(acomp->dev); + } + + return 0; +} +EXPORT_SYMBOL_GPL(hda_display_power); + +int hda_get_display_clk(struct hdac_bus *bus) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + return acomp->ops->get_cdclk_freq(acomp->dev); +} +EXPORT_SYMBOL_GPL(hda_get_display_clk); + +static int hda_component_master_bind(struct device *dev) +{ + struct i915_audio_component *acomp = hda_acomp; + int ret; + + ret = component_bind_all(dev, acomp); + if (ret < 0) + return ret; + + if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && + acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { + ret = -EINVAL; + goto out_unbind; + } + + /* + * Atm, we don't support dynamic unbinding initiated by the child + * component, so pin its containing module until we unbind. + */ + if (!try_module_get(acomp->ops->owner)) { + ret = -ENODEV; + goto out_unbind; + } + + return 0; + +out_unbind: + component_unbind_all(dev, acomp); + + return ret; +} + +static void hda_component_master_unbind(struct device *dev) +{ + struct i915_audio_component *acomp = hda_acomp; + + module_put(acomp->ops->owner); + component_unbind_all(dev, acomp); + WARN_ON(acomp->ops || acomp->dev); +} + +static const struct component_master_ops hda_component_master_ops = { + .bind = hda_component_master_bind, + .unbind = hda_component_master_unbind, +}; + +static int hda_component_master_match(struct device *dev, void *data) +{ + /* i915 is the only supported component */ + return !strcmp(dev->driver->name, "i915"); +} + +int hda_i915_init(struct hdac_bus *bus) +{ + struct component_match *match = NULL; + struct device *dev = bus->dev; + struct i915_audio_component *acomp; + int ret; + + acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); + if (!acomp) + return -ENOMEM; + bus->audio_component = acomp; + hda_acomp = acomp; + + component_match_add(dev, &match, hda_component_master_match, bus); + ret = component_master_add_with_match(dev, &hda_component_master_ops, + match); + if (ret < 0) + goto out_err; + + /* + * Atm, we don't support deferring the component binding, so make sure + * i915 is loaded and that the binding successfully completes. + */ + request_module("i915"); + + if (!acomp->ops) { + ret = -ENODEV; + goto out_master_del; + } + dev_dbg(dev, "bound to i915 component master\n"); + + return 0; +out_master_del: + component_master_del(dev, &hda_component_master_ops); +out_err: + kfree(acomp); + bus->audio_component = NULL; + dev_err(dev, "failed to add i915 component master (%d)\n", ret); + + return ret; +} +EXPORT_SYMBOL_GPL(hda_i915_init); + +int hda_i915_exit(struct hdac_bus *bus) +{ + struct device *dev = bus->dev; + struct i915_audio_component *acomp = bus->audio_component; + + WARN_ON(bus->i915_power_refcount); + if (bus->i915_power_refcount > 0 && acomp && acomp->ops) + acomp->ops->put_power(acomp->dev); + + component_master_del(dev, &hda_component_master_ops); + + kfree(acomp); + bus->audio_component = NULL; + + return 0; +} +EXPORT_SYMBOL_GPL(hda_i915_exit); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 117bf5c..98ced97 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -145,11 +145,6 @@ config SND_HDA_CODEC_HDMI comment "Set to Y if you want auto-loading the codec driver" depends on SND_HDA=y && SND_HDA_CODEC_HDMI=m -config SND_HDA_I915 - bool - default y - depends on DRM_I915 - config SND_HDA_CODEC_CIRRUS tristate "Build Cirrus Logic codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index c5e6651..a32ab83 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,7 +1,5 @@ snd-hda-intel-objs := hda_intel.o snd-hda-tegra-objs := hda_tegra.o -# for haswell power well -snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o snd-hda-codec-y := hda_bind.o hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o snd-hda-codec-y += hda_controller.o diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c deleted file mode 100644 index 8521702..0000000 --- a/sound/pci/hda/hda_i915.c +++ /dev/null @@ -1,227 +0,0 @@ -/* - * hda_i915.c - routines for Haswell HDA controller power well support - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the Free - * Software Foundation; either version 2 of the License, or (at your option) - * any later version. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY - * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License - * for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software Foundation, - * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/pci.h> -#include <linux/component.h> -#include <drm/i915_component.h> -#include <sound/core.h> -#include "hda_controller.h" -#include "hda_intel.h" - -/* Intel HSW/BDW display HDA controller Extended Mode registers. - * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display - * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled. - */ -#define AZX_REG_EM4 0x100c -#define AZX_REG_EM5 0x1010 - -int hda_set_codec_wakeup(struct hda_intel *hda, bool enable) -{ - struct i915_audio_component *acomp = &hda->audio_component; - - if (!acomp->ops) - return -ENODEV; - - if (!acomp->ops->codec_wake_override) { - dev_warn(&hda->chip.pci->dev, - "Invalid codec wake callback\n"); - return 0; - } - - dev_dbg(&hda->chip.pci->dev, "%s codec wakeup\n", - enable ? "enable" : "disable"); - - acomp->ops->codec_wake_override(acomp->dev, enable); - - return 0; -} - -int hda_display_power(struct hda_intel *hda, bool enable) -{ - struct i915_audio_component *acomp = &hda->audio_component; - - if (!acomp->ops) - return -ENODEV; - - dev_dbg(&hda->chip.pci->dev, "display power %s\n", - enable ? "enable" : "disable"); - - if (enable) { - if (!hda->i915_power_refcount++) - acomp->ops->get_power(acomp->dev); - } else { - WARN_ON(!hda->i915_power_refcount); - if (!--hda->i915_power_refcount) - acomp->ops->put_power(acomp->dev); - } - - return 0; -} - -void haswell_set_bclk(struct hda_intel *hda) -{ - int cdclk_freq; - unsigned int bclk_m, bclk_n; - struct i915_audio_component *acomp = &hda->audio_component; - struct pci_dev *pci = hda->chip.pci; - - /* Only Haswell/Broadwell need set BCLK */ - if (pci->device != 0x0a0c && pci->device != 0x0c0c - && pci->device != 0x0d0c && pci->device != 0x160c) - return; - - if (!acomp->ops) - return; - - cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev); - switch (cdclk_freq) { - case 337500: - bclk_m = 16; - bclk_n = 225; - break; - - case 450000: - default: /* default CDCLK 450MHz */ - bclk_m = 4; - bclk_n = 75; - break; - - case 540000: - bclk_m = 4; - bclk_n = 90; - break; - - case 675000: - bclk_m = 8; - bclk_n = 225; - break; - } - - azx_writew(&hda->chip, EM4, bclk_m); - azx_writew(&hda->chip, EM5, bclk_n); -} - -static int hda_component_master_bind(struct device *dev) -{ - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - struct i915_audio_component *acomp = &hda->audio_component; - int ret; - - ret = component_bind_all(dev, acomp); - if (ret < 0) - return ret; - - if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && - acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { - ret = -EINVAL; - goto out_unbind; - } - - /* - * Atm, we don't support dynamic unbinding initiated by the child - * component, so pin its containing module until we unbind. - */ - if (!try_module_get(acomp->ops->owner)) { - ret = -ENODEV; - goto out_unbind; - } - - return 0; - -out_unbind: - component_unbind_all(dev, acomp); - - return ret; -} - -static void hda_component_master_unbind(struct device *dev) -{ - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - struct i915_audio_component *acomp = &hda->audio_component; - - module_put(acomp->ops->owner); - component_unbind_all(dev, acomp); - WARN_ON(acomp->ops || acomp->dev); -} - -static const struct component_master_ops hda_component_master_ops = { - .bind = hda_component_master_bind, - .unbind = hda_component_master_unbind, -}; - -static int hda_component_master_match(struct device *dev, void *data) -{ - /* i915 is the only supported component */ - return !strcmp(dev->driver->name, "i915"); -} - -int hda_i915_init(struct hda_intel *hda) -{ - struct component_match *match = NULL; - struct device *dev = &hda->chip.pci->dev; - struct i915_audio_component *acomp = &hda->audio_component; - int ret; - - component_match_add(dev, &match, hda_component_master_match, hda); - ret = component_master_add_with_match(dev, &hda_component_master_ops, - match); - if (ret < 0) - goto out_err; - - /* - * Atm, we don't support deferring the component binding, so make sure - * i915 is loaded and that the binding successfully completes. - */ - request_module("i915"); - - if (!acomp->ops) { - ret = -ENODEV; - goto out_master_del; - } - - dev_dbg(dev, "bound to i915 component master\n"); - - return 0; -out_master_del: - component_master_del(dev, &hda_component_master_ops); -out_err: - dev_err(dev, "failed to add i915 component master (%d)\n", ret); - - return ret; -} - -int hda_i915_exit(struct hda_intel *hda) -{ - struct device *dev = &hda->chip.pci->dev; - struct i915_audio_component *acomp = &hda->audio_component; - - WARN_ON(hda->i915_power_refcount); - if (hda->i915_power_refcount > 0 && acomp->ops) - acomp->ops->put_power(acomp->dev); - - component_master_del(dev, &hda_component_master_ops); - - return 0; -} diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 706879a..229a1e4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -57,6 +57,8 @@ #endif #include <sound/core.h> #include <sound/initval.h> +#include <sound/hdaudio.h> +#include <sound/hda_i915.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -493,13 +495,13 @@ static void azx_init_pci(struct azx *chip) static void hda_intel_init_chip(struct azx *chip, bool full_reset) { - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - hda_set_codec_wakeup(hda, true); + hda_set_codec_wakeup(bus, true); azx_init_chip(chip, full_reset); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - hda_set_codec_wakeup(hda, false); + hda_set_codec_wakeup(bus, false); } /* calculate runtime delay from LPIB */ @@ -557,9 +559,9 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) /* Enable/disable i915 display power for the link */ static int azx_intel_link_power(struct azx *chip, bool enable) { - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); - return hda_display_power(hda, enable); + return hda_display_power(bus, enable); } /* @@ -797,6 +799,50 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ +/* Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK + * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) + * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: + * BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled and need to + * be restored to avoid abnormal playback speed. + */ +static void haswell_set_bclk(struct hda_intel *hda) +{ + struct azx *chip = &hda->chip; + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!hda->need_i915_power) + return; + + cdclk_freq = hda_get_display_clk(azx_bus(chip)); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } + + azx_writew(chip, HSW_EM4, bclk_m); + azx_writew(chip, HSW_EM5, bclk_n); +} + #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -830,7 +876,7 @@ static int azx_suspend(struct device *dev) pci_disable_msi(chip->pci); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) - hda_display_power(hda, false); + hda_display_power(bus, false); return 0; } @@ -851,7 +897,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { - hda_display_power(hda, true); + hda_display_power(azx_bus(chip), true); haswell_set_bclk(hda); } if (chip->msi) @@ -895,7 +941,7 @@ static int azx_runtime_suspend(struct device *dev) azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) - hda_display_power(hda, false); + hda_display_power(azx_bus(chip), false); return 0; } @@ -905,6 +951,7 @@ static int azx_runtime_resume(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; struct hda_intel *hda; + struct hdac_bus *bus; struct hda_codec *codec; int status; @@ -921,11 +968,12 @@ static int azx_runtime_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { - hda_display_power(hda, true); + bus = azx_bus(chip); + hda_display_power(bus, true); haswell_set_bclk(hda); /* toggle codec wakeup bit for STATESTS read */ - hda_set_codec_wakeup(hda, true); - hda_set_codec_wakeup(hda, false); + hda_set_codec_wakeup(bus, true); + hda_set_codec_wakeup(bus, false); } /* Read STATESTS before controller reset */ @@ -1143,10 +1191,11 @@ static int azx_free(struct azx *chip) #ifdef CONFIG_SND_HDA_PATCH_LOADER release_firmware(chip->fw); #endif + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { if (hda->need_i915_power) - hda_display_power(hda, false); - hda_i915_exit(hda); + hda_display_power(bus, false); + hda_i915_exit(bus); } kfree(hda); @@ -1905,6 +1954,7 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { static int azx_probe_continue(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; int dev = chip->dev_index; int err; @@ -1921,11 +1971,11 @@ static int azx_probe_continue(struct azx *chip) if (pci->device != 0x0f04 && pci->device != 0x2284) hda->need_i915_power = 1; - err = hda_i915_init(hda); + err = hda_i915_init(bus); if (err < 0) goto i915_power_fail; - err = hda_display_power(hda, true); + err = hda_display_power(bus, true); if (err < 0) { dev_err(chip->card->dev, "Cannot turn on display power on i915\n"); @@ -1977,7 +2027,7 @@ static int azx_probe_continue(struct azx *chip) out_free: if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && !hda->need_i915_power) - hda_display_power(hda, false); + hda_display_power(bus, false); i915_power_fail: if (err < 0) diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 7fd3254..354f0bb 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -16,7 +16,6 @@ #ifndef __SOUND_HDA_INTEL_H #define __SOUND_HDA_INTEL_H -#include <drm/i915_component.h> #include "hda_controller.h" struct hda_intel { @@ -44,36 +43,7 @@ struct hda_intel { /* secondary power domain for hdmi audio under vga device */ struct dev_pm_domain hdmi_pm_domain; - /* i915 component interface */ bool need_i915_power:1; /* the hda controller needs i915 power */ - struct i915_audio_component audio_component; - int i915_power_refcount; }; -#ifdef CONFIG_SND_HDA_I915 -int hda_set_codec_wakeup(struct hda_intel *hda, bool enable); -int hda_display_power(struct hda_intel *hda, bool enable); -void haswell_set_bclk(struct hda_intel *hda); -int hda_i915_init(struct hda_intel *hda); -int hda_i915_exit(struct hda_intel *hda); -#else -static inline int hda_set_codec_wakeup(struct hda_intel *hda, bool enable) -{ - return 0; -} -static inline int hda_display_power(struct hda_intel *hda, bool enable) -{ - return 0; -} -static inline void haswell_set_bclk(struct hda_intel *hda) { return; } -static inline int hda_i915_init(struct hda_intel *hda) -{ - return 0; -} -static inline int hda_i915_exit(struct hda_intel *hda) -{ - return 0; -} -#endif - #endif -- 1.9.1

3 2

[alsa-devel] Low volume since linux 3.19
by Jan Hinnerk Stosch 18 May '15

18 May '15

Hi, I hope this is the right place to report my issue!? Since linux 3.19 I have very low volume on my Fujitsu E753 headphone even on highest volume level. As I don't have much experience with audio configuration and it worked before I did a kernel bisect and found out, that commit "03ad6a8c93b6df2d65c305b5b5f9474068b45bfb" is "responsible" for this. Is there something that I have to change in my configuration? Or do you need some further information? Here is some output from dmesg: [ 5.853009] sound hdaudioC0D0: ALC269VB: SKU not ready 0x909701f0 [ 5.853857] sound hdaudioC0D0: autoconfig: line_outs=1 (0x21/0x0/0x0/0x0/0x0) type:line [ 5.853860] sound hdaudioC0D0: speaker_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 5.853862] sound hdaudioC0D0: hp_outs=1 (0x1a/0x0/0x0/0x0/0x0) [ 5.853864] sound hdaudioC0D0: mono: mono_out=0x0 [ 5.853866] sound hdaudioC0D0: inputs: [ 5.853869] sound hdaudioC0D0: Mic=0x18 [ 5.853871] sound hdaudioC0D0: Dock Mic=0x1b [ 5.853873] sound hdaudioC0D0: Internal Mic=0x12 Thanks in advance, jhs

2 6

[alsa-devel] [PATCH 1/3] ALSA: hda - add hdac stream trace
by libin.yang＠intel.com 18 May '15

18 May '15

From: Libin Yang <libin.yang(a)intel.com> Add the trace of snd_hdac_stream_start and snd_hdac_stream_stop. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/hda/hdac_stream.c | 5 +++++ sound/hda/trace.h | 27 +++++++++++++++++++++++++++ 2 files changed, 32 insertions(+) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 1ba0462..52a894f 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -10,6 +10,7 @@ #include <sound/pcm.h> #include <sound/hdaudio.h> #include <sound/hda_register.h> +#include "trace.h" /** * snd_hdac_stream_init - initialize each stream (aka device) @@ -48,6 +49,8 @@ void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start) { struct hdac_bus *bus = azx_dev->bus; + trace_snd_hdac_stream_start(bus, azx_dev); + azx_dev->start_wallclk = snd_hdac_chip_readl(bus, WALLCLK); if (!fresh_start) azx_dev->start_wallclk -= azx_dev->period_wallclk; @@ -82,6 +85,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_clear); */ void snd_hdac_stream_stop(struct hdac_stream *azx_dev) { + trace_snd_hdac_stream_stop(azx_dev->bus, azx_dev); + snd_hdac_stream_clear(azx_dev); /* disable SIE */ snd_hdac_chip_updatel(azx_dev->bus, INTCTL, 1 << azx_dev->index, 0); diff --git a/sound/hda/trace.h b/sound/hda/trace.h index 33a7eb5..e27e2c0 100644 --- a/sound/hda/trace.h +++ b/sound/hda/trace.h @@ -50,6 +50,33 @@ TRACE_EVENT(hda_unsol_event, ), TP_printk("%s", __get_str(msg)) ); + +DECLARE_EVENT_CLASS(hdac_stream, + TP_PROTO(struct hdac_bus *bus, struct hdac_stream *azx_dev), + + TP_ARGS(bus, azx_dev), + + TP_STRUCT__entry( + __field(unsigned char, stream_tag) + ), + + TP_fast_assign( + __entry->stream_tag = (azx_dev)->stream_tag; + ), + + TP_printk("stream_tag: %d", __entry->stream_tag) +); + +DEFINE_EVENT(hdac_stream, snd_hdac_stream_start, + TP_PROTO(struct hdac_bus *bus, struct hdac_stream *azx_dev), + TP_ARGS(bus, azx_dev) +); + +DEFINE_EVENT(hdac_stream, snd_hdac_stream_stop, + TP_PROTO(struct hdac_bus *bus, struct hdac_stream *azx_dev), + TP_ARGS(bus, azx_dev) +); + #endif /* __HDAC_TRACE_H */ /* This part must be outside protection */ -- 1.9.1

2 3

[alsa-devel] [PATCH 1/6] ALSA: hda/tegra - Set CORBRP self-clear flag
by Thierry Reding 18 May '15

18 May '15

From: Thierry Reding <treding(a)nvidia.com> This is set for the MCP variants of the NVIDIA HDA controller, which the Tegra variant was derived from. This fixes the following warning at boot time: [ 2.486610] tegra-hda 70030000.hda: CORB reset timeout#1, CORBRP = 0 Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index db0bb50fa5b9..7003677f7473 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -460,11 +460,12 @@ MODULE_DEVICE_TABLE(of, hda_tegra_match); static int hda_tegra_probe(struct platform_device *pdev) { + const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY | + AZX_DCAPS_CORBRP_SELF_CLEAR; struct snd_card *card; struct azx *chip; struct hda_tegra *hda; int err; - const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY; hda = devm_kzalloc(&pdev->dev, sizeof(*hda), GFP_KERNEL); if (!hda) -- 2.3.5

2 7

Re: [alsa-devel] Support Dell headset mode for ALC298 platform
by Takashi Iwai 18 May '15

18 May '15

At Mon, 18 May 2015 07:38:53 +0000, Kailang wrote: > > Hi Takashi, > > Dell has new platform for ALC298. > Attach patch was enable headset mode for ALC298 machine. Applied, thanks. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-support-dell-headset-mode-for-alc298.patch <application/octet-stream (base64)>] >

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - ALC292 dock fix for Thinkpad L450
by Ansgar Hegerfeld 18 May '15

18 May '15

The Lenovo ThinkPad L450 requires the ALC292_FIXUP_TPT440_DOCK fix in order to get sound output on the docking stations audio port. This patch was tested using a ThinkPad L450 (20DSS00B00) using kernel 4.0.3 and a ThinkPad Pro Dock. Signed-off-by: Ansgar Hegerfeld <linux(a)hegerfeld.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e2afd53..a9f41a4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5148,6 +5148,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),

2 1

[alsa-devel] [PATCH] ALSA: hda - Add Conexant codecs CX20721, CX20722, CX20723 and CX20724
by David Henningsson 18 May '15

18 May '15

This patch adds support for Conexant HD Audio codecs CX20721, CX20722, CX20723 and CX20724. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1454656 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f8f0dfb..78b719b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -968,6 +968,14 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f150b9, .name = "CX20665", .patch = patch_conexant_auto }, + { .id = 0x14f150f1, .name = "CX20721", + .patch = patch_conexant_auto }, + { .id = 0x14f150f2, .name = "CX20722", + .patch = patch_conexant_auto }, + { .id = 0x14f150f3, .name = "CX20723", + .patch = patch_conexant_auto }, + { .id = 0x14f150f4, .name = "CX20724", + .patch = patch_conexant_auto }, { .id = 0x14f1510f, .name = "CX20751/2", .patch = patch_conexant_auto }, { .id = 0x14f15110, .name = "CX20751/2", @@ -1002,6 +1010,10 @@ MODULE_ALIAS("snd-hda-codec-id:14f150ab"); MODULE_ALIAS("snd-hda-codec-id:14f150ac"); MODULE_ALIAS("snd-hda-codec-id:14f150b8"); MODULE_ALIAS("snd-hda-codec-id:14f150b9"); +MODULE_ALIAS("snd-hda-codec-id:14f150f1"); +MODULE_ALIAS("snd-hda-codec-id:14f150f2"); +MODULE_ALIAS("snd-hda-codec-id:14f150f3"); +MODULE_ALIAS("snd-hda-codec-id:14f150f4"); MODULE_ALIAS("snd-hda-codec-id:14f1510f"); MODULE_ALIAS("snd-hda-codec-id:14f15110"); MODULE_ALIAS("snd-hda-codec-id:14f15111"); -- 1.9.1

2 1

[alsa-devel] [PATCH 0/4] Fixes and cleanups for the recent procfs changes
by Takashi Iwai 18 May '15

18 May '15

Hi, here is a series of patches to fix and clean up the recent changes for ALSA core procfs support code. Takashi === Takashi Iwai (4): ALSA: info: Fix leaks of child entries at snd_info_free_entry() ALSA: info: Register proc entries recursively, too ALSA: info: Move list addition to snd_info_create_entry() ALSA: info: Drop kerneldoc comment from snd_info_create_entry() sound/core/info.c | 62 +++++++++++++++++++++++++++++++++++++++++-------------- sound/core/init.c | 18 ++++++---------- 2 files changed, 52 insertions(+), 28 deletions(-) -- 2.3.6

1 4

[alsa-devel] [PATCH 1/1] ALSA: core: Fix regression for ELD/codec information files
by han.lu＠intel.com 18 May '15

18 May '15

From: "Lu, Han" <han.lu(a)intel.com> The commit c560a6797e3b ("ALSA: core: Remove child proc file elements recursively") introduced a regression that ELD/codec information files can not be found on /proc/asound/cardX/. This patch corrects it. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/sound/info.h b/include/sound/info.h index 1626995..ae8d30a 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -149,6 +149,10 @@ static inline int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp) { *entryp = snd_info_create_card_entry(card, name, card->proc_root); + if (snd_info_register(*entryp) < 0) { + snd_info_free_entry(*entryp); + *entryp = NULL; + } return *entryp ? 0 : -ENOMEM; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix headset mic and mic-in for a Dell desktop
by David Henningsson 18 May '15

18 May '15

ALC662 does not need any special verbs to change the jack functionality, and enables mic in through the headphone jack mode by changing the direction of the headphone pin node. BugLink: https://bugs.launchpad.net/bugs/1454235 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 46 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 45 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce41271..30ec30f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3674,6 +3674,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, alc_process_coef_fw(codec, coef0293); snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); break; + case 0x10ec0662: + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; case 0x10ec0668: alc_write_coef_idx(codec, 0x11, 0x0001); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); @@ -4012,7 +4016,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) if (new_headset_mode != ALC_HEADSET_MODE_MIC) { snd_hda_set_pin_ctl_cache(codec, hp_pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); - if (spec->headphone_mic_pin) + if (spec->headphone_mic_pin && spec->headphone_mic_pin != hp_pin) snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin, PIN_VREFHIZ); } @@ -4215,6 +4219,18 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec, } } +static void alc_fixup_headset_mode_alc662(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + spec->gen.hp_mic = 1; /* Mic-in is same pin as headphone */ + } else + alc_fixup_headset_mode(codec, fix, action); +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6080,7 +6096,9 @@ enum { ALC662_FIXUP_NO_JACK_DETECT, ALC662_FIXUP_ZOTAC_Z68, ALC662_FIXUP_INV_DMIC, + ALC662_FIXUP_DELL_MIC_NO_PRESENCE, ALC668_FIXUP_DELL_MIC_NO_PRESENCE, + ALC662_FIXUP_HEADSET_MODE, ALC668_FIXUP_HEADSET_MODE, ALC662_FIXUP_BASS_MODE4_CHMAP, ALC662_FIXUP_BASS_16, @@ -6273,6 +6291,20 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE }, + [ALC662_FIXUP_DELL_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a1113c }, /* use as headset mic, without its own jack detect */ + /* headphone mic by setting pin control of 0x1b (headphone out) to in + vref_50 */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_HEADSET_MODE + }, + [ALC662_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc662, + }, [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -6432,6 +6464,18 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {0x1f, 0x411111f0} static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { + SND_HDA_PIN_QUIRK(0x10ec0662, 0x1028, "Dell", ALC662_FIXUP_DELL_MIC_NO_PRESENCE, + {0x12, 0x4004c000}, + {0x14, 0x01014010}, + {0x15, 0x411111f0}, + {0x16, 0x411111f0}, + {0x18, 0x01a19020}, + {0x19, 0x411111f0}, + {0x1a, 0x0181302f}, + {0x1b, 0x0221401f}, + {0x1c, 0x411111f0}, + {0x1d, 0x4054c601}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, ALC668_STANDARD_PINS, {0x12, 0x99a30130}, -- 1.9.1

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.1
by Mark Brown 18 May '15

18 May '15

The following changes since commit 030bbdbf4c833bc69f502eae58498bc5572db736: Linux 4.1-rc3 (2015-05-10 15:12:29 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.1-rc3 for you to fetch changes up to 7730c0b55043f6ff1c27e9cb45e13679995b2361: Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/mc13783', 'asoc/fix/uda1380', 'asoc/fix/wm8960' and 'asoc/fix/wm8994' into asoc-linus (2015-05-12 20:31:33 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.1 A few more fixes for v4.1, some driver fixes plus one core fix which fixes registration of DAI links when adding prefixes to CODECs to deuplicate in multi-CODEC systems. ---------------------------------------------------------------- Axel Lin (2): ASoC: mc13783: Fix wrong mask value used in mc13xxx_reg_rmw() calls ASoC: rt5645: Fix mask for setting RT5645_DMIC_2_DP_GPIO12 bit Fang, Yang A (1): ASoC: rt5645: Add ACPI match ID Koro Chen (1): ASoC: dapm: Modify widget stream name according to prefix Mark Brown (3): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/mc13783', 'asoc/fix/uda1380', 'asoc/fix/wm8960' and 'asoc/fix/wm8994' into asoc-linus Peter Ujfalusi (1): ASoC: davinci-mcasp: Correct pm status check in suspend callback Vasily Khoruzhick (1): ASoC: uda1380: Avoid accessing i2c bus when codec is disabled Zidan Wang (2): ASoC: wm8960: fix "RINPUT3" audio route error ASoC: wm8994: correct BCLK DIV 348 to 384 sound/soc/codecs/mc13783.c | 4 ++-- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/soc-dapm.c | 11 ++++++++--- 6 files changed, 14 insertions(+), 9 deletions(-)

2 1

Re: [alsa-devel] Fwd: Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 17 May '15

17 May '15

> > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/patch/sound/pci > > /hda/patch_realtek.c?id=c3e837bbcc03e39991da5dd1f791c06cde9f7a45 > > > > When the static model were used, the jack was configured as line out > > instead of headphone jack and there is no headphone playback volume > > That's rather "relevant change". Good job in finding it. I would like such > configuration (or better) back. > I don't see any side effect neither with your changes to hda_generic nor > with > vanilla kernel, but I don't know what problems should I expect. > Do headphone get same volume level when the jack is configured as Line Out (without enable Headphone Amplifier) ? The driver use different automute method when the jack is configured as HP and Line Out https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… if you look at speaker.conf , lineout.conf and headphone.conf, it seem pulseaudio is unlikely mute all internal speakers when one pair of the playback volume control need to be shared by internal speaker and external speakers you need dirver 's automute instead of pulseaudio 's automute ==> Best config: lo_type=2, wired=1, mio=1 multi_outs = 15/0/0/0 : 2/3/4/0 (type HP) out path: depth=3 '02:0c:15' multi_ios(2) = 1a/18 : 3/4 mio path: depth=3 '03:0d:1a' mio path: depth=3 '04:0e:18' spk_outs = 14/17/16/0 : 5/25/4/0 spk path: depth=3 '05:0f:14' spk path: depth=3 '25:26:17' spk path: depth=3 '04:0e:16' https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… when line out type == AUTO_PIN_HP_OUT should first do_mute() use speaker_path instead of out_path <https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p…> >>> I started MS Windows and without externals speakers plugged in I >> >> could test >> > 5.1 configuration and it seemed that sound was coming from >> different directions. >> >> > >> > What I managed to determine: >> > I've got 3 stacks: >> > 1 ST: Headphone, Line-out (FL and FR), SPDIF >> > 2 ST: Microphone, Line-out (CLFE) >> > 3 ST: Line-in, Line-out (RL and RR) >> > >> > I've got 5 responsive output pins >> > PIN 0x14: Internal FL FR >> > PIN 0x15: External FL FR >> > PIN 0x17: Internal LFE >> > PIN 0x18: External CLFE >> > PIN 0x1a: External RL RR >> > >> > How can we make it work with these information? speaker-test -Dhw:0,0 -c6 -t wav >> >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/Document… >> [1] >> >> seem chnmap type _FIXED , _VAR and _PAIRED does not fit when LFE and >> FC can be swapped for internal speaker and external speaker > > > Right, you can't do it with a single map of one of those types. http://git.alsa-project.org/?p=alsa-utils.git&a=search&h=HEAD&st=commit&s=c… do chmap of internal speakers need {FL FR RL RR LFE NA} ? http://git.alsa-project.org/?p=alsa-lib.git&a=search&h=HEAD&st=commit&s=chm… as set_chmap() still not yet implemented in pcm_route.c do speaker-test need to specify channel map when using those odd number channels ? speaker-test -D surround21 -c3 -t wav speaker-test -D surround41 -c5 -t wav -mFL,FR,RL,RR,LFE speaker-test -D surround50 -c5 -t wav http://git.alsa-project.org/?p=alsa-utils.git;a=commitdiff;h=7b14c00b56784f… this example seem not working speaker-test -c2 -t wav -Dplug:front -mFR,FL > > >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> [2] >> >> How do the driver implement channel map when codec has swap >> center/lfe switch ? >> >> Do chmap contain both {FL FR RL RR FC LFE} and {FL FR RL RR LFE FC} ? > > > If you want to provide support for userspace doing it both ways, then yes, you need those two _FIXED maps. Can chmap contain two _FIXED maps of same number of channels ?

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[alsa-devel] Microphone issues with Microsoft LifeCam Studio
by Dainius Masiliūnas 16 May '15

16 May '15

I'm having an issue with the Microsoft LifeCam Studio in that the camera does not work if the integrated microphone in the camera is activated at the same time. When that happens, the kernel log outputs a variety of warnings such as: cannot get freq at ep 0x82 ALSA lib pcm_dsnoop.c:614:(snd_pcm_dsnoop_open) unable to open slave uvcvideo: Failed to query (GET_DEF) UVC control 2 on unit 4: -32 (exp. 2). uvcvideo: Failed to query (GET_DEF) UVC control 2 on unit 4: 0 (exp. 2). ALSA lib pcm.c:7980:(snd_pcm_set_params) Channels count (1) not available for CAPTURE: Incorrect argument ALSA lib pcm.c:7956:(snd_pcm_set_params) Broken configuration for CAPTURE: no configurations available ALSA lib pcm_route.c:867:(find_matching_chmap) Found no matching channel map uvcvideo: Failed to set UVC probe control : -32 (exp. 26). There is a workaround in that disabling the built-in microphone by blacklisting snd_usb_audio does make the camera work, but obviously it's not ideal at all, as that way half the camera's functionality is not available and no other USB audio peripherals can be used. I'm assuming that this is some sort of a quirk in the camera itself; the UVC page ( http://www.ideasonboard.org/uvc/#footnote-13 ) does mention that there is an issue with the bandwidth requests, but setting the 0x80 quirk does not change the situation with the microphone in any way, unfortunately. I am running the latest mainline kernel (kernel-ml from elrepo) on CentOS 7.1. Can anything be done about this issue, and is there anything I could do to help debug it? Also, this issue has been noted a few years back, but it was left unsolved: https://bugs.launchpad.net/ubuntu/+source/guvcview/+bug/1033146 https://sourceforge.net/p/linux-uvc/mailman/message/29645043/ The original reporter said it was a Flash bug, but that is in no way the case because my machine does not have Flash installed to begin with (I'm testing this with WebRTC). -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] [pulseaudio-discuss] ThinkPad T-510 audio output mute LED non-workingness
by Takashi Iwai 16 May '15

16 May '15

[Switched Cc to alsa-devel ML, as it's irrelevant from PA but rather a driver issue.] At Mon, 27 Apr 2015 07:58:59 -0600, Glenn Golden wrote: > > Takashi Iwai <tiwai(a)suse.de> [2015-04-27 08:50:34 +0200]: > > > > Gah, I hate reports without alsa-info.sh output (not a link to > > somewhere else). > > > > Glenn, please attach alsa-info.sh outputs (run it with --no-upload > > option) and the whole dmesg output after boot. 3.19 should already > > contain the hook to thinkpad_acpi.c. If anything is missing, you > > should have a warning indicating it. > > > > Apologies for the omissions. Both are attached. Thanks! > Fwiw -- perhaps nothing at this point -- a good deal of info has also been > posted to the kernel bugtracker ticket: > > https://bugzilla.kernel.org/show_bug.cgi?id=96171 > > Re: "...you should have a warning indicating it": I'm an unskilled laboror > here w.r.t. kernel affairs, and not even sure what sort of warnings to be > looking for, so may have missed something important without realizing it. > Just let me know what add'l info you need, I'll be happy to post it. I do > appreciate your attention to the issue. I see no such warning in the log you attached, so it's not the issue, as it seems. Below is a debug patch, and now you should see messages like "XXX ..." in kernel log at loading the driver. If there is no such a message, it means that the thinkpad hook isn't built in at all. Please check the kernel log again with the patch. Takashi --- diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 0a4ad5feb82e..48ce61e46599 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -34,6 +34,7 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) if (old_vmaster_hook) old_vmaster_hook(private_data, enabled); + pr_info("XXX mute led updated %d\n", enabled); if (led_set_func) led_set_func(TPACPI_LED_MUTE, !enabled); } @@ -58,8 +59,10 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + codec_info(codec, "XXX hda_fixup_thinkpad_acpi init...\n"); if (!is_thinkpad(codec)) return; + codec_info(codec, "XXX ok, this is thinkpad\n"); if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -70,18 +73,24 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, removefunc = true; if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { + codec_info(codec, "XXX ok, mute LED is hooked\n"); old_vmaster_hook = spec->vmaster_mute.hook; spec->vmaster_mute.hook = update_tpacpi_mute_led; removefunc = false; + } else { + codec_err(codec, "XXX mute LED NOT WORKING!\n"); } if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { if (spec->num_adc_nids > 1) codec_dbg(codec, "Skipping micmute LED control due to several ADCs"); else { + codec_info(codec, "XXX ok, mic LED is hooked\n"); spec->cap_sync_hook = update_tpacpi_micmute_led; removefunc = false; } + } else { + codec_err(codec, "XXX mic LED NOT WORKING!\n"); } }

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