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

[alsa-devel] [PATCH] ASoC: rk3399_gru_sound: fix recording pop at first attempt
by Xing Zheng 22 Sep '16

22 Sep '16

From: Wonjoon Lee <woojoo.lee(a)samsung.com> Pop happens when mclk applied but dmic's own boot-time Specify dmic delay times in dt to make sure clocks are ready earlier than dmic working Signed-off-by: Wonjoon Lee <woojoo.lee(a)samsung.com> Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- .../bindings/sound/rockchip,rk3399-gru-sound.txt | 6 ++++++ sound/soc/rockchip/rk3399_gru_sound.c | 14 ++++++++++++++ 2 files changed, 20 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt index f19b6c8..b7dd3ab 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -6,6 +6,12 @@ Required properties: connected to the codecs - rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs +Optional properties: +- dmic-delay : specify delay time for DMIC ready. + If this option is specified, which means it's required dmic need + delay for DMIC to ready so that rt5514 can avoid recording before + DMIC send valid data + Example: sound { diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 164b6da..6ab838b 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -37,6 +37,8 @@ #define SOUND_FS 256 +unsigned int rt5514_dmic_delay; + static struct snd_soc_jack rockchip_sound_jack; static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { @@ -122,6 +124,9 @@ static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, return ret; } + /* Wait for DMIC stable */ + msleep(rt5514_dmic_delay); + return 0; } @@ -334,6 +339,15 @@ static int rockchip_sound_probe(struct platform_device *pdev) return -ENODEV; } + /* Set DMIC delay */ + ret = device_property_read_u32(&pdev->dev, "dmic-delay", + &rt5514_dmic_delay); + if (ret) { + rt5514_dmic_delay = 0; + dev_dbg(&pdev->dev, + "no optional property 'dmic-delay' found, default: no delay\n"); + } + rockchip_dailinks[DAILINK_RT5514_DSP].cpu_name = kstrdup_const(dev_name(dev), GFP_KERNEL); rockchip_dailinks[DAILINK_RT5514_DSP].cpu_dai_name = kstrdup_const(dev_name(dev), GFP_KERNEL); rockchip_dailinks[DAILINK_RT5514_DSP].platform_name = kstrdup_const(dev_name(dev), GFP_KERNEL); -- 1.7.9.5

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[alsa-devel] USB composite audio device problem with newer kernel versions
by Pecsi, Gergely 22 Sep '16

22 Sep '16

Hi all, We are having issues with a special multiple interface audio device. The card has 3 devices with the same VID/PID, 2 of them is a mono output device and the third one has an input and an output channel also. This third device does not function properly with newer kernel versions, the audio capture is not working. Tests: -our Debian Jessie based distro with kernel 3.16 -> OK -our Debian Jessie based distro with kernel 4.1.25 -> OK -our Debian Jessie based distro with kernel 4.4.1 -> NOK -our Debian Jessie based distro with kernel 4.6 -> NOK -our Debian Jessie based distro with kernel 4.7 -> NOK -Debian testing daily snapshot with kernel 4.6.0 -> NOK In the NOK cases, when we try to capture the stream with arecord, we get: "audio open error: No such file or directory". We can also see that in the NOK cases the arecord labels the capture device with device number 0, but in the OK cases it labels it with device number 2. Aplay labels these devices with device number 2 in any case. Arecord list output in the good case: card 5: Module [Spc Module], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 and in the bad case: card 5: Module [Spc Module], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 Relevant parts of alsa-info.sh output attached. What could be the root of the problem? What areas in the alsa codebase could be involved? Should we report an ALSA bug? Best regards, Gergely Pecsi

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[alsa-devel] Oxygen Possible Bug in FCM
by xerofoify 22 Sep '16

22 Sep '16

Greetings All, Do anyone known of the best way to test this function, ac97_fp_rec_volume_put. Seems there may be a bug in it. I do have the hardware so that's not a issue just wondering what the different between fp_rec and the standard get functions are in terms of testing. Further more I am assuming this has to do with the fp headphones switch but I don't have headphones that are fp so is there another way to test this particular function without checking sound levels. Cheers, Nick

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[alsa-devel] [PATCH] ALSA: snd-usb-line6 depends on CONFIG_SND_HWDEP
by Valdis Kletnieks 22 Sep '16

22 Sep '16

ALSA - snd-usb-line6 depends on CONFIG_SND_HWDEP ERROR: "snd_hwdep_new" [sound/usb/line6/snd-usb-line6.ko] undefined! scripts/Makefile.modpost:91: recipe for target '__modpost' failed make[1]: *** [__modpost] Error 1 Signed-off-by: Valdis Kletnieks <valdis.kletnieks(a)vt.edu> diff --git a/sound/usb/line6/Kconfig b/sound/usb/line6/Kconfig index 8ffcf48097d9..39b400392d71 100644 --- a/sound/usb/line6/Kconfig +++ b/sound/usb/line6/Kconfig @@ -2,6 +2,7 @@ config SND_USB_LINE6 tristate select SND_RAWMIDI select SND_PCM + select SND_HWDEP config SND_USB_POD tristate "Line 6 POD USB support"

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[alsa-devel] usb: Novation Twitch not working on Linux 4.7
by Juan Pedro Bolívar Puente 21 Sep '16

21 Sep '16

Hi! I am trying to use again my Novation Twitch after some time, and audio playback is not working anymore. I have tried with Jackd and various other programs, and it seems to always fail to install the hardware params. Note that before it fails, I do get to hear a little noise in the monitor output. Here is console the output of various programs. Note that, I am using Linux 4.7.0-1-amd64 as distributed with Debian Sid. Here is the output of some commands: $ ./alsacap *** Scanning for playback devices *** ... Card 1, ID `Twitch', name `Twitch' Device 0, ID `USB Audio', name `USB Audio', 1 subdevices (1 available) 4 channels, sampling rate 44100..48000 Hz Sample formats: S24_3LE Subdevice 0, name `subdevice #0' $ ./alsacap -d hw:Twitch *** Exploring configuration space of device `hw:Twitch' for playback *** 4 channels Sampling rate 44100..48000 Hz Sample formats: S24_3LE Significant bits: 24 $ aplay -D hw:Twitch -f S24_3LE -c 4 -r 44100 test.wav Playing raw data 'prueba.wav' : Signed 24 bit Little Endian in 3bytes, Rate 44100 Hz, Channels 4 aplay: set_params:1361: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S24_3LE SUBFORMAT: STD SAMPLE_BITS: 24 FRAME_BITS: 96 CHANNELS: 4 RATE: 44100 PERIOD_TIME: (125011 125012) PERIOD_SIZE: 5513 PERIOD_BYTES: 66156 PERIODS: (3 4) BUFFER_TIME: 500000 BUFFER_SIZE: 22050 BUFFER_BYTES: 264600 TICK_TIME: 0 $ mplayer -ao alsa:device=hw=Twitch test.mp3 MPlayer 1.3.0 (Debian), built with gcc-5.4.0 (C) 2000-2016 MPlayer Team Playing test.mp3. ... [AO_ALSA] Format s16le is not supported by hardware, trying default. [AO_ALSA] Unable to set format: Invalid argument Failed to initialize audio driver 'alsa:device=hw=Twitch' $ mplayer -ao alsa:device=plughw=Twitch test.mp3 MPlayer 1.3.0 (Debian), built with gcc-5.4.0 (C) 2000-2016 MPlayer Team Playing test.mp3. ... [AO_ALSA] alsa-lib: pcm_hw.c:603:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE failed (-32): Broken pipe [AO_ALSA] alsa-lib: pcm_direct.c:1054:(snd1_pcm_direct_initialize_slave) unable to install hw params [AO_ALSA] alsa-lib: pcm_dmix.c:1053:(snd_pcm_dmix_open) unable to initialize slave [AO_ALSA] Playback open error: Broken pipe Failed to initialize audio driver 'alsa:device=hw=Twitch' $ jackd -n twitch -d alsa -d hw:Twitch jackdmp 1.9.11 ... creating alsa driver ... hw:Twitch|hw:Twitch|1024|2|48000|0|0|nomon|swmeter|-|32bit Using ALSA driver USB-Audio running on card 1 - Novation Twitch at usb-0000:00:1d.0-1.2, full speed ALSA: Cannot open PCM device alsa_pcm for capture. Falling back to playback-only mode configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 periods ALSA: final selected sample format for playback: 24bit little-endian in 3bytes format ALSA: use 2 periods for playback ALSA: cannot set hardware parameters for playback ALSA: cannot configure playback channel Am I doing something wrong? Is it really broken? Thanks a lot! JP -- ∿∿∿∿ https://sinusoid.es ∿∿∿∿

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[alsa-devel] [alsa-lib][PATCH] ucm: docs: typeset lists of identifiers explicitly
by Antonio Ospite 21 Sep '16

21 Sep '16

Doxygen doesn't preserve formatting that relies only on indentation, typeset lists of identifiers explicitly. This makes the HTML docs a lot more readable. This change comes along the lines of commit 72aa0f8332fb ("ucm: reformat snd_use_case_get() doc"). Some TABs has been added too in order to preserve the aligned look when reading the source code. Signed-off-by: Antonio Ospite <ao2(a)ao2.it> --- include/use-case.h | 57 +++++++++++++++++++++++++++--------------------------- 1 file changed, 29 insertions(+), 28 deletions(-) diff --git a/include/use-case.h b/include/use-case.h index 9aac6e2..1ebd945 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -192,20 +192,21 @@ int snd_use_case_free_list(const char *list[], int items); * \return Number of list entries if success, otherwise a negative error code * * Defined identifiers: - * NULL - get card list - * (in pair cardname+comment) - * _verbs - get verb list - * (in pair verb+comment) - * _devices[/{verb}] - get list of supported devices - * (in pair device+comment) - * _modifiers[/{verb}]- get list of supported modifiers - * (in pair modifier+comment) - * TQ[/{verb}] - get list of TQ identifiers - * _enadevs - get list of enabled devices - * _enamods - get list of enabled modifiers + * - NULL - get card list + * (in pair cardname+comment) + * - _verbs - get verb list + * (in pair verb+comment) + * - _devices[/{verb}] - get list of supported devices + * (in pair device+comment) + * - _modifiers[/{verb}] - get list of supported modifiers + * (in pair modifier+comment) + * - TQ[/{verb}] - get list of TQ identifiers + * - _enadevs - get list of enabled devices + * - _enamods - get list of enabled modifiers + * + * - _supporteddevs/{modifier}|{device}[/{verb}] - list of supported devices + * - _conflictingdevs/{modifier}|{device}[/{verb}] - list of conflicting devices * - * _supporteddevs/{modifier}|{device}[/{verb}] - list of supported devices - * _conflictingdevs/{modifier}|{device}[/{verb}] - list of conflicting devices * Note that at most one of the supported/conflicting devs lists has * any entries, and when neither is present, all devices are supported. * @@ -331,8 +332,8 @@ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, * \return Zero if success, otherwise a negative error code * * Known identifiers: - * _devstatus/{device} - return status for given device - * _modstatus/{modifier} - return status for given modifier + * - _devstatus/{device} - return status for given device + * - _modstatus/{modifier} - return status for given modifier */ int snd_use_case_geti(snd_use_case_mgr_t *uc_mgr, const char *identifier, @@ -346,19 +347,19 @@ int snd_use_case_geti(snd_use_case_mgr_t *uc_mgr, * \return Zero if success, otherwise a negative error code * * Known identifiers: - * _verb - set current verb = value - * _enadev - enable given device = value - * _disdev - disable given device = value - * _swdev/{old_device} - new_device = value - * - disable old_device and then enable new_device - * - if old_device is not enabled just return - * - check transmit sequence firstly - * _enamod - enable given modifier = value - * _dismod - disable given modifier = value - * _swmod/{old_modifier} - new_modifier = value - * - disable old_modifier and then enable new_modifier - * - if old_modifier is not enabled just return - * - check transmit sequence firstly + * - _verb - set current verb = value + * - _enadev - enable given device = value + * - _disdev - disable given device = value + * - _swdev/{old_device} - new_device = value + * - disable old_device and then enable new_device + * - if old_device is not enabled just return + * - check transmit sequence firstly + * - _enamod - enable given modifier = value + * = _dismod - disable given modifier = value + * - _swmod/{old_modifier} - new_modifier = value + * - disable old_modifier and then enable new_modifier + * - if old_modifier is not enabled just return + * - check transmit sequence firstly */ int snd_use_case_set(snd_use_case_mgr_t *uc_mgr, const char *identifier, -- 2.9.3

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[alsa-devel] [PATCH -next] ASoC: sti: fix missing clk_disable_unprepare() on error in uni_player_start()
by Wei Yongjun 21 Sep '16

21 Sep '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Fix the missing clk_disable_unprepare() before return from uni_player_start() in the error handling case. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sti/uniperif_player.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 645e415..1bc8ebc 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -893,8 +893,10 @@ static int uni_player_start(struct uniperif *player) SET_UNIPERIF_SOFT_RST_SOFT_RST(player); ret = reset_player(player); - if (ret < 0) + if (ret < 0) { + clk_disable_unprepare(player->clk); return ret; + } /* * Does not use IEC61937 features of the uniperipheral hardware.

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[alsa-devel] [PATCH -next] ASoC: rt5663: fix sparse warnings
by Wei Yongjun 21 Sep '16

21 Sep '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Fixes the following sparse warnings: sound/soc/codecs/rt5663.c:1367:14: warning: duplicate const sound/soc/codecs/rt5663.c:1577:5: warning: symbol 'rt5663_button_detect' was not declared. Should it be static? sound/soc/codecs/rt5663.c:2857:24: warning: symbol 'rt5663_aif_dai_ops' was not declared. Should it be static? sound/soc/codecs/rt5663.c:2866:27: warning: symbol 'rt5663_dai' was not declared. Should it be static? sound/soc/codecs/rt5663.c:3193:6: warning: symbol 'rt5663_i2c_shutdown' was not declared. Should it be static? sound/soc/codecs/rt5663.c:3200:19: warning: symbol 'rt5663_i2c_driver' was not declared. Should it be static? Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt5663.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 7963a27..fa79568 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1364,7 +1364,7 @@ static const char * const rt5663_if1_adc_data_select[] = { "L/R", "R/L", "L/L", "R/R" }; -static const SOC_ENUM_SINGLE_DECL(rt5663_if1_adc_enum, RT5663_TDM_2, +static SOC_ENUM_SINGLE_DECL(rt5663_if1_adc_enum, RT5663_TDM_2, RT5663_DATA_SWAP_ADCDAT1_SHIFT, rt5663_if1_adc_data_select); static void rt5663_enable_push_button_irq(struct snd_soc_codec *codec, @@ -1574,7 +1574,7 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) return rt5663->jack_type; } -int rt5663_button_detect(struct snd_soc_codec *codec) +static int rt5663_button_detect(struct snd_soc_codec *codec) { int btn_type, val; @@ -2854,7 +2854,7 @@ static int rt5663_resume(struct snd_soc_codec *codec) #define RT5663_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt5663_aif_dai_ops = { +static struct snd_soc_dai_ops rt5663_aif_dai_ops = { .hw_params = rt5663_hw_params, .set_fmt = rt5663_set_dai_fmt, .set_sysclk = rt5663_set_dai_sysclk, @@ -2863,7 +2863,7 @@ struct snd_soc_dai_ops rt5663_aif_dai_ops = { .set_bclk_ratio = rt5663_set_bclk_ratio, }; -struct snd_soc_dai_driver rt5663_dai[] = { +static struct snd_soc_dai_driver rt5663_dai[] = { { .name = "rt5663-aif", .id = RT5663_AIF, @@ -3190,14 +3190,14 @@ static int rt5663_i2c_remove(struct i2c_client *i2c) return 0; } -void rt5663_i2c_shutdown(struct i2c_client *client) +static void rt5663_i2c_shutdown(struct i2c_client *client) { struct rt5663_priv *rt5663 = i2c_get_clientdata(client); regmap_write(rt5663->regmap, RT5663_RESET, 0); } -struct i2c_driver rt5663_i2c_driver = { +static struct i2c_driver rt5663_i2c_driver = { .driver = { .name = "rt5663", .owner = THIS_MODULE,

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[alsa-devel] [snd_ali5451] snd_ali5451 0000:00:06.0: BUG: , pos = 16384, buffer size = 16384, period size = 1024
by Enrico Mioso 21 Sep '16

21 Sep '16

Hello guys. The message on the subject comes from my dmesg and is generated by the said module when using mplayer. (MPlayer SVN-r37379 (C) 2000-2015 MPlayer Team) Including some devices from my lspci: if you wish to see the integral one, let me know. 00:06.0 Multimedia audio controller: ULi Electronics Inc. M5451 PCI AC-Link Controller Audio Device (rev 02) Subsystem: Hewlett-Packard Company Device 002a Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR+ <PERR+ INTx- Latency: 64 (500ns min, 6000ns max) Interrupt: pin A routed to IRQ 5 Region 0: I/O ports at 1000 [size=256] Region 1: Memory at d0001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [dc] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: snd_ali5451 00:07.0 ISA bridge: ULi Electronics Inc. M1533/M1535/M1543 PCI to ISA Bridge [Aladdin IV/V/V+] Subsystem: ULi Electronics Inc. ALi M1533 Aladdin IV/V ISA Bridge Control: I/O+ Mem+ BusMaster+ SpecCycle+ MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Capabilities: [a0] Power Management version 1 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- 00:08.0 Modem: ULi Electronics Inc. M5457 AC'97 Modem Controller (prog-if 00 [Generic]) Subsystem: Hewlett-Packard Company Device 002a Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin A routed to IRQ 10 Region 0: Memory at d0002000 (32-bit, non-prefetchable) [size=4K] Region 1: I/O ports at 1400 [size=256] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Thank you. P.S.: sorry for giving no follow-up to my previous hd-audio thing; thanks.

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[alsa-devel] USB Gadget audio not working
by noman pouigt 21 Sep '16

21 Sep '16

Hello, I am trying to use my device as a sound card and enabled CONFIG_USB_AUDIO. However I don't know what kind of device node should be added in the machine file? I just want to record whatever is coming into the USB played back by the host over USB. I could get my device to be detected as a sound card i.e. USB_AUDIO_CLASS in my ubuntu machine. My understanding of USB audio is very less so I thought below change should be sufficient. Below are the only changes which i have done to enable USB_AUDIO. Machine file: { .name = "something_random", .stream_name = "USB Capture", .cpu_dai_name = "snd-soc-dummy-dai", .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .ignore_suspend = 1, }, drivers/usb/gadget/u_uac1.c : #define FILE_PCM_PLAYBACK "/dev/snd/pcmC0D0p" #define FILE_PCM_CAPTURE "/dev/snd/pcmC0D0c" in defconfig CONFIG_USB_AUDIO=y Please let me know if I am doing anything wrong. Thanks,

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