June 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoc: wm8731: add 32bit mode.
by Matt Flax 17 Jun '16

17 Jun '16

This patch adds 32 bit word capability to the wm8731 driver. The wm8731 codec is capable of handling 32 bit word sizes, however that has not previously been activated in the codec driver. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/wm8731.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 4bcf5f8..d18261a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -358,6 +358,9 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, case 24: iface |= 0x0008; break; + case 32: + iface |= 0x000c; + break; } wm8731_set_deemph(codec); @@ -541,7 +544,7 @@ static int wm8731_startup(struct snd_pcm_substream *substream, #define WM8731_RATES SNDRV_PCM_RATE_8000_96000 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8731_dai_ops = { .startup = wm8731_startup, -- 2.7.4

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[alsa-devel] [PATCH 0/3] ASoC: ak4613/wm8940: Regmap cache fixes
by Geert Uytterhoeven 17 Jun '16

17 Jun '16

Hi Liam, Mark, This series contains 3 fixes for regmap cache handling in the ak4613 and wm8940 drivers. All fixes are based on recent similar fixes for the ak4642 driver. The first patch fixes a crash I've seen on r8a7795/Salvator-X. The last patch is completely untested. The issue was discovered using 'git grep -Lw cache_type -- $(git grep -lw regcache_sync -- $(git grep -lw regmap_config))'' Thanks! Geert Uytterhoeven (3): ASoC: ak4613: Enable cache usage to fix crashes on resume ASoC: ak4613: Implement suspend callback ASoC: wm8940: Enable cache usage to fix crashes on resume sound/soc/codecs/ak4613.c | 13 ++++++++++++- sound/soc/codecs/wm8940.c | 1 + 2 files changed, 13 insertions(+), 1 deletion(-) -- 1.9.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[alsa-devel] [PATCH] snd: soc: cx20442 codec: set tty->receiver_room in v253_open
by Janusz Krzysztofik 17 Jun '16

17 Jun '16

Commit 79901317ce80 ("n_tty: Don't flush buffer when closing ldisc"), introduced in v3.10, revealed a bug in the cx20442 codec driver which has never been setting tty->receive_room on line discipline open as it should from the beginning. Fix it. Created and tested on Amstrad Delta against Linux-4.7-rc3 Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- sound/soc/codecs/cx20442.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d6f4abb..fb3885f 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -226,6 +226,7 @@ static int v253_open(struct tty_struct *tty) if (!tty->disc_data) return -ENODEV; + tty->receive_room = 16; if (tty->ops->write(tty, v253_init, len) != len) { ret = -EIO; goto err; -- 2.7.3

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[alsa-devel] [PATCH] ASoC: codecs: Remove "select REGMAP_SPI/REGMAP_I2C"
by Alexander Shiyan 17 Jun '16

17 Jun '16

REGMAP_SPI and REGMAP_I2C is selected automatically by SND_SOC, so no additional actions is required. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/Kconfig | 21 --------------------- 1 file changed, 21 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4d82a58..1e03fa0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -288,12 +288,10 @@ config SND_SOC_ADAU1761 config SND_SOC_ADAU1761_I2C tristate select SND_SOC_ADAU1761 - select REGMAP_I2C config SND_SOC_ADAU1761_SPI tristate select SND_SOC_ADAU1761 - select REGMAP_SPI config SND_SOC_ADAU1781 select SND_SOC_ADAU17X1 @@ -302,12 +300,10 @@ config SND_SOC_ADAU1781 config SND_SOC_ADAU1781_I2C tristate select SND_SOC_ADAU1781 - select REGMAP_I2C config SND_SOC_ADAU1781_SPI tristate select SND_SOC_ADAU1781 - select REGMAP_SPI config SND_SOC_ADAU1977 tristate @@ -315,12 +311,10 @@ config SND_SOC_ADAU1977 config SND_SOC_ADAU1977_SPI tristate select SND_SOC_ADAU1977 - select REGMAP_SPI config SND_SOC_ADAU1977_I2C tristate select SND_SOC_ADAU1977 - select REGMAP_I2C config SND_SOC_ADAV80X tristate @@ -403,7 +397,6 @@ config SND_SOC_CS42L73 config SND_SOC_CS4265 tristate "Cirrus Logic CS4265 CODEC" depends on I2C - select REGMAP_I2C # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 @@ -425,13 +418,11 @@ config SND_SOC_CS4271_I2C tristate "Cirrus Logic CS4271 CODEC (I2C)" depends on I2C select SND_SOC_CS4271 - select REGMAP_I2C config SND_SOC_CS4271_SPI tristate "Cirrus Logic CS4271 CODEC (SPI)" depends on SPI_MASTER select SND_SOC_CS4271 - select REGMAP_SPI config SND_SOC_CS42XX8 tristate @@ -440,7 +431,6 @@ config SND_SOC_CS42XX8_I2C tristate "Cirrus Logic CS42448/CS42888 CODEC (I2C)" depends on I2C select SND_SOC_CS42XX8 - select REGMAP_I2C # Cirrus Logic CS4349 HiFi DAC config SND_SOC_CS4349 @@ -580,13 +570,11 @@ config SND_SOC_PCM3168A_I2C tristate "Texas Instruments PCM3168A CODEC - I2C" depends on I2C select SND_SOC_PCM3168A - select REGMAP_I2C config SND_SOC_PCM3168A_SPI tristate "Texas Instruments PCM3168A CODEC - SPI" depends on SPI_MASTER select SND_SOC_PCM3168A - select REGMAP_SPI config SND_SOC_PCM5102A tristate @@ -598,13 +586,11 @@ config SND_SOC_PCM512x_I2C tristate "Texas Instruments PCM512x CODECs - I2C" depends on I2C select SND_SOC_PCM512x - select REGMAP_I2C config SND_SOC_PCM512x_SPI tristate "Texas Instruments PCM512x CODECs - SPI" depends on SPI_MASTER select SND_SOC_PCM512x - select REGMAP_SPI config SND_SOC_RL6231 tristate @@ -668,7 +654,6 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate - select REGMAP_I2C select REGMAP_IRQ config SND_SOC_RT5677_SPI @@ -715,13 +700,11 @@ config SND_SOC_SSM2602_SPI tristate "Analog Devices SSM2602 CODEC - SPI" depends on SPI_MASTER select SND_SOC_SSM2602 - select REGMAP_SPI config SND_SOC_SSM2602_I2C tristate "Analog Devices SSM2602 CODEC - I2C" depends on I2C select SND_SOC_SSM2602 - select REGMAP_I2C config SND_SOC_SSM4567 tristate "Analog Devices ssm4567 amplifier driver support" @@ -730,7 +713,6 @@ config SND_SOC_SSM4567 config SND_SOC_STA32X tristate "STA326, STA328 and STA329 speaker amplifier" depends on I2C - select REGMAP_I2C config SND_SOC_STA350 tristate "STA350 speaker amplifier" @@ -788,7 +770,6 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC31XX tristate "Texas Instruments TLV320AIC31xx CODECs" depends on I2C - select REGMAP_I2C config SND_SOC_TLV320AIC32X4 tristate @@ -918,13 +899,11 @@ config SND_SOC_WM8804_I2C tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C" depends on I2C select SND_SOC_WM8804 - select REGMAP_I2C config SND_SOC_WM8804_SPI tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI" depends on SPI_MASTER select SND_SOC_WM8804 - select REGMAP_SPI config SND_SOC_WM8900 tristate -- 2.4.9

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[alsa-devel] [alsa-lib][PATCH 0/9 v3] ctl: add APIs for control element set
by Takashi Sakamoto 16 Jun '16

16 Jun '16

Hi, This patchset is a revised version of below one: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-June/109041.html Changes: - Drop a patch for deprecating old APIs. - In detail, see our discussion: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-June/109117.html - Revise comments for TLV feature. - I realized it better to use vendor-neutral representation, after reading this discussion: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069483.h… Again, ALSA developers in PulseAudio side, test program in the last patch causes an abort to pulseaudio process as a result of kicking assertion of hcontrol in alsa-lib. In detail, please refer to my previous patchset. Regards Takashi Sakamoto (9): ctl: add an overview for design of ALSA control interface ctl: improve comments for handling element data ctl: add functions to add an element set ctl: improve comments for API to add an element of IEC958 type ctl: change former APIs as wrapper functions of element set APIs pcm: use new APIs to add a control element set for softvol plugin ctl: add explanation about threshold level feature ctl: improve API documentation for threshold level operations ctl: add test program for control element set include/control.h | 21 ++ src/control/control.c | 813 ++++++++++++++++++++++++++++++++------------ src/pcm/pcm_softvol.c | 7 +- test/Makefile.am | 5 +- test/user-ctl-element-set.c | 561 ++++++++++++++++++++++++++++++ 5 files changed, 1189 insertions(+), 218 deletions(-) create mode 100644 test/user-ctl-element-set.c -- 2.7.4

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[alsa-devel] [PATCH v3 0/2] ASoC: Add support to Qualcomm msm8916-wcd codec
by Srinivas Kandagatla 16 Jun '16

16 Jun '16

This patchset aims at adding msm8916-wcd codec support. msm8916-wcd codec is found in Qualcomm msm8916 and apq8016 processors. This codec IP is split in to two parts(Digital & Analog), Analog part is integrated in to PMIC PM8916 and the digital part is integrated into Application processor. Register access to the analog part is done via SPMI interface to PMIC, and registers on the Application processor are memory mapped. Data transfer between Analog and Digital Die is done via a internal bus called PDM. Most of this driver is ported from downstream Andriod v3.10 kernel. Jack support and Headset button detection code is not added in to this series. This codec support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary Mic. - 2 Digital Microphones. - 2 Mic Bias Circuits. - Earpiece - Headset - Loud Speaker. - Jack Detect. - Headset Button detection circuit. Tested this code on v4.6-rc2 with lpass driver capture support patches for Headset (Mic and speakers), Secondary Mic and Loudspeaker on DB410C. DMIC should work too. Thanks, srini Changes since RFC v2: - Fixed error handling on regluator_enable suggested by Kenneth. - Fixed compatible strings suggested by Kenneth - make max/min rates inline with actual supported rates, spotted by Kenneth - Few minor suggestions by Kenneth. Changes since RFC v1 (https://lkml.org/lkml/2016/2/16/710) - Lot of code cleanup as suggest by Mark Brown. - Fixed bindings as suggested by Mark Rutland. - merged all the patches to one as spliting the patches did not help review. Srinivas Kandagatla (2): ASoC: msm8916: Add codec Device Tree bindings. ASoC: msm8916: Add msm8916-wcd codec driver .../devicetree/bindings/sound/qcom,msm8916-wcd.txt | 103 ++ include/dt-bindings/sound/msm8916-wcd.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 +- sound/soc/codecs/msm8916-wcd-registers.h | 710 +++++++++ sound/soc/codecs/msm8916-wcd.c | 1575 ++++++++++++++++++++ sound/soc/codecs/msm8916-wcd.h | 308 ++++ 7 files changed, 2709 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd.txt create mode 100644 include/dt-bindings/sound/msm8916-wcd.h create mode 100644 sound/soc/codecs/msm8916-wcd-registers.h create mode 100644 sound/soc/codecs/msm8916-wcd.c create mode 100644 sound/soc/codecs/msm8916-wcd.h -- 2.8.3

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Re: [alsa-devel] ASoC: rt5514: add rt5514 SPI driver
by Dan Carpenter 16 Jun '16

16 Jun '16

Hello Oder Chiou, This is a semi-automatic email about new static checker warnings. The patch 6eebf35b0e4a: "ASoC: rt5514: add rt5514 SPI driver" from Jun 6, 2016, leads to the following Smatch complaint: sound/soc/codecs/rt5514-spi.c:87 rt5514_spi_copy_work() warn: variable dereferenced before check 'rt5514_dsp->substream' (see line 83) sound/soc/codecs/rt5514-spi.c 82 container_of(work, struct rt5514_dsp, copy_work.work); 83 struct snd_pcm_runtime *runtime = rt5514_dsp->substream->runtime; ^^^^^^^^^^^^^^^^^^^^^^^ Dereference. 84 size_t period_bytes, truncated_bytes = 0; 85 86 mutex_lock(&rt5514_dsp->dma_lock); 87 if (!rt5514_dsp->substream) { ^^^^^^^^^^^^^^^^^^^^^ Checked too late. 88 dev_err(rt5514_dsp->dev, "No pcm substream\n"); 89 goto done; regards, dan carpenter

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[alsa-devel] [CX2075] No sound after suspend/resume from front speaker
by Max Zhao 16 Jun '16

16 Jun '16

Hi, I have recently been using Archlinux with standard kernel 4.6.2 on a Toshiba Z20T-B (Intel Broadwell). After using suspend to RAM (S3) no sound is output from the laptop speakers. (systemctl suspend) The headphone jack behaves normally. Even after suspend. Restarting pulseaudio, killing it etc does not work. Pulseaudio registers no errors and tries to output sound to speakers as normal. aplay -D front:PCH also reports no errors, but fails silently. Suspend on Win10 works with sound resume, so it should not be a hardware or UEFI bug. Best regards, Max Zhao lspci -vvnn 00:1b.0 Audio device [0403]: Intel Corporation Wildcat Point-LP High Definition Audio Controller [8086:9ca0] (rev 03) Subsystem: Toshiba America Info Systems Device [1179:0001] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 49 Region 0: Memory at f1234000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel 00:03.0 Audio device [0403]: Intel Corporation Broadwell-U Audio Controller [8086:160c] (rev 09) Subsystem: Toshiba America Info Systems Device [1179:0001] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 50 Region 0: Memory at f1230000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel 00:1b.0 should be the main audio card. 00:03.0 controls hdmi output. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: CX20753/4 Analog [CX20753/4 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default ALSA Output (currently PulseAudio Sound Server) hdmi:CARD=HDMI,DEV=0 HDA Intel HDMI, HDMI 0 HDMI Audio Output hdmi:CARD=HDMI,DEV=1 HDA Intel HDMI, HDMI 1 HDMI Audio Output hdmi:CARD=HDMI,DEV=2 HDA Intel HDMI, HDMI 2 HDMI Audio Output sysdefault:CARD=PCH HDA Intel PCH, CX20753/4 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers amixer -c1 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 74 Mono: Playback 65 [88%] [-9.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 0 [0%] [-74.00dB] [off] Front Right: Playback 0 [0%] [-74.00dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Beep',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 7 Mono: Playback 4 [57%] [-12.00dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 80 Front Left: Capture 80 [100%] [6.00dB] [on] Front Right: Capture 80 [100%] [6.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [24.00dB] Front Right: 2 [67%] [24.00dB] /proc/asound/card1/codec#0 Codec: Conexant CX20753/4 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f15111 Subsystem Id: 0x11790694 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold S3D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20753/4 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=0, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] Converter: stream=0, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x12 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x04] Node 0x13 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Device: name="CX20753/4 Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x50 0x50] [0x50 0x50] [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 4 0x18 0x1a* 0x19 0x1e Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 4 0x19* 0x1a 0x15 0x1e Node 0x15 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x16 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x04211040: [Jack] HP Out at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x17 [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000010: OUT Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00001124: IN Detect Vref caps: HIZ 80 Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00001124: IN Detect Vref caps: HIZ 80 Pin Default 0x04a11030: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00000020: IN Pin Default 0x95a60120: [Fixed] Mic at Int Top Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x1e [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000024: IN Detect Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0

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Re: [alsa-devel] [PATCH v5] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 16 Jun '16

16 Jun '16

// + Adam, On 2016年06月16日 03:43, Steve Barber wrote: > On Mon, Jun 13, 2016 at 1:42 AM, Xing Zheng<zhengxing(a)rock-chips.com> wrote: >> + >> +static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, >> + struct snd_pcm_hw_params *params) >> +{ >> + struct snd_soc_pcm_runtime *rtd = substream->private_data; >> + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; >> + struct snd_soc_dai *codec_dai = rtd->codec_dai; >> + struct snd_soc_dapm_context *dapm =&rtd->card->dapm; >> + int mclk, ret; >> + >> + snd_soc_dapm_disable_pin(dapm, "Speakers"); >> + snd_soc_dapm_disable_pin(dapm, "Int Mic"); >> + snd_soc_dapm_enable_pin(dapm, "Headphones"); >> + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); >> + snd_soc_dapm_sync(dapm); >> + >> + /* in bypass mode, the mclk has to be one of the frequencies below */ >> + switch (params_rate(params)) { >> + case 8000: >> + case 16000: >> + case 24000: >> + case 32000: >> + case 48000: >> + case 64000: >> + case 96000: >> + mclk = 12288000; >> + break; >> + case 11025: >> + case 22050: >> + case 44100: >> + case 88200: >> + mclk = 11289600; >> + break; >> + default: >> + return -EINVAL; >> + } >> + >> + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, >> + SND_SOC_CLOCK_OUT); >> + if (ret< 0) { >> + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); >> + return ret; >> + } >> + >> + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, >> + SND_SOC_CLOCK_IN); >> + if (ret< 0) { >> + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); >> + return ret; >> + } >> + >> + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); >> + if (ret< 0) { >> + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); >> + return ret; >> + } >> + >> + return 0; >> +} >> + >> +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) >> +{ >> + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; >> + int ret; >> + >> + /* Enable Headset and 4 Buttons Jack detection */ >> + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", >> + SND_JACK_HEADSET | > Should this also include SND_JACK_LINEOUT? da7219 differentiates > between the two so not including lineout means, for example, we can > miss jack insertion/removal events. > Hi Adam, Could you please confirm Steve says that about SND_JACK_LINEOUT? Thanks. -- - Xing Zheng

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[alsa-devel] [PATCH] ALSA: hda: fix some klockwork scan warnings
by libin.yang＠linux.intel.com 16 Jun '16

16 Jun '16

From: Libin Yang <libin.yang(a)linux.intel.com> This patch fixes some warnings from klockwork. These warnings are not the real issues. The patch adds the sanity check. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/hda/hdmi_chmap.c | 28 +++++++++++++++++++++++++--- sound/pci/hda/hda_codec.c | 8 +++++++- sound/pci/hda/patch_hdmi.c | 5 +++++ 3 files changed, 37 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index c6c75e7..81acc20 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -353,7 +353,8 @@ static void hdmi_std_setup_channel_mapping(struct hdac_chmap *chmap, int hdmi_slot = 0; /* fill actual channel mappings in ALSA channel (i) order */ for (i = 0; i < ch_alloc->channels; i++) { - while (!ch_alloc->speakers[7 - hdmi_slot] && !WARN_ON(hdmi_slot >= 8)) + while (!WARN_ON(hdmi_slot >= 8) && + !ch_alloc->speakers[7 - hdmi_slot]) hdmi_slot++; /* skip zero slots */ hdmi_channel_mapping[ca][i] = (i << 4) | hdmi_slot++; @@ -430,6 +431,12 @@ static int to_cea_slot(int ordered_ca, unsigned char pos) int mask = snd_hdac_chmap_to_spk_mask(pos); int i; + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (ordered_ca >= ARRAY_SIZE(channel_allocations)) + return -1; + if (mask) { for (i = 0; i < 8; i++) { if (channel_allocations[ordered_ca].speakers[7 - i] == mask) @@ -456,7 +463,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_spk_to_chmap); /* from CEA slot to ALSA API channel position */ static int from_cea_slot(int ordered_ca, unsigned char slot) { - int mask = channel_allocations[ordered_ca].speakers[7 - slot]; + int mask; + + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (slot >= 8) + return 0; + + mask = channel_allocations[ordered_ca].speakers[7 - slot]; return snd_hdac_spk_to_chmap(mask); } @@ -523,7 +538,8 @@ static void hdmi_setup_fake_chmap(unsigned char *map, int ca) int ordered_ca = get_channel_allocation_order(ca); for (i = 0; i < 8; i++) { - if (i < channel_allocations[ordered_ca].channels) + if (ordered_ca < ARRAY_SIZE(channel_allocations) && + i < channel_allocations[ordered_ca].channels) map[i] = from_cea_slot(ordered_ca, hdmi_channel_mapping[ca][i] & 0x0f); else map[i] = 0; @@ -551,6 +567,12 @@ int snd_hdac_get_active_channels(int ca) { int ordered_ca = get_channel_allocation_order(ca); + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (ordered_ca >= ARRAY_SIZE(channel_allocations)) + ordered_ca = 0; + return channel_allocations[ordered_ca].channels; } EXPORT_SYMBOL_GPL(snd_hdac_get_active_channels); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8374188..9913be8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3584,6 +3584,12 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, bool reset; spdif = snd_hda_spdif_out_of_nid(codec, nid); + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (WARN_ON(spdif == NULL)) + return; + curr_fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); reset = codec->spdif_status_reset && @@ -3768,7 +3774,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, spdif = snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid); if (mout->dig_out_nid && mout->share_spdif && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { - if (chs == 2 && + if (chs == 2 && spdif != NULL && snd_hda_is_supported_format(codec, mout->dig_out_nid, format) && !(spdif->status & IEC958_AES0_NONAUDIO)) { diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d0d5ad8..56e5204 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1680,6 +1680,11 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) mutex_lock(&codec->spdif_mutex); spdif = snd_hda_spdif_out_of_nid(codec, cvt_nid); + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (WARN_ON(spdif == NULL)) + return true; non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); mutex_unlock(&codec->spdif_mutex); return non_pcm; -- 1.9.1

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