September 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] motu microbook iic
by thomas rost 26 Feb '20

26 Feb '20

Hello, can anybody help me get the 'class compliant' motu microbook iic to run? this is my alsa-info.sh output: http://alsa-project.org/db/?f=0fd9a8df7bae3400d83b6af8b75c45aa0760d84d thanks in advance. thomas

2 1

[alsa-devel] [PATCH v9] ASoC: pxa: switch to new ac97 bus support
by Robert Jarzmik 24 Feb '20

24 Feb '20

Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v8: fixed the trivial compilation error v7 was in https://patchwork.kernel.org/patch/9951919/ --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++--- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++++++---------------------- 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..f1f25704fe52 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -36,7 +36,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484ab3c2ad67..1a0b55beb282 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -20,13 +20,12 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S tristate diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1b41c0f2a8fb..80e4757176d8 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -219,6 +212,9 @@ static const struct snd_soc_component_driver pxa_ac97_component = { static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -231,10 +227,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -245,8 +245,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- 2.11.0

2 1

[alsa-devel] ALSA 1.0.29 release
by Jaroslav Kysela 19 Feb '20

19 Feb '20

Hello all, all userspace packages (except alsa-oss) were released today. You may download them from the ALSA website http://www.alsa-project.org or directly from the FTP server ftp://ftp.alsa-project.org/pub . Full list of changes: http://www.alsa-project.org/main/index.php/Changes_v1.0.28_v1.0.29 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

2 16

[alsa-devel] [PATCH V6 0/4] update supported sample format
by Shengjiu Wang 18 Jan '20

18 Jan '20

This patch serial is to update the supported format for fsl_asrc and fix some format issue Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch changes in v3 - Fix build report by kbuild test robot <lkp(a)intel.com> - change snd_dmaengine_pcm_set_runtime_hwparams to snd_dmaengine_pcm_refine_runtime_hwparams changes in v4 - update according to Nicolin's comments. changes in v5 - free asrc pair when error happens in patch 4/4 changes in v6 - ignore return value of snd_soc_set_runtime_hwparams in patch 4/4 include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 64 ++++++++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 61 ++------------------ 6 files changed, 199 insertions(+), 86 deletions(-) -- 2.21.0

5 13

[alsa-devel] Simultaneous play/record on implicit feedback device causes 'endpoint in use' error
by Robert Giaraffa 15 Jan '20

15 Jan '20

My company (Silicon Labs) makes a CP2615 USB audio class 1.0 device which uses implicit feedback. I saw these two issues when testing with aplay/arecord/alsabat/alsaloop: * During playback-only operation, the implicit feedback (IN) endpoint was not active. * During play/record operation, the implicit feedback values were not acted on, i.e. the host always sent 192 bytes per OUT packet even if previous IN packets had been 188 or 196 bytes. This behavior was observed on these systems: RaspberryPi2: Raspbian 9 (stretch), kernel v4.14.72 x86-64: Ubuntu 18.04, kernel v4.15.0-36 I was able to get implicit feedback working properly in playback mode by adding a device-specific quirk to 'pcm.c' that is similar to existing quirks for implicit-feedback devices, e.g. M-Audio Fast Track Ultra. (Note: The following patch applies to kernel v4.14.72. In v4.18, the set_sync_ep_implicit_fb_quirk function was changed so that there is a single call to usb_ifnum_to_if(), rather than a call for each device-dependent quirk.) $ git diff pcm.c diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b9c9a19f..6fb5567b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -358,6 +358,21 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x10C4, 0xEAC1): // Silicon Labs CP2615 + // Ensure CP2615 is configured in Async Mode + if (attr != USB_ENDPOINT_SYNC_ASYNC) + return 0; + + // In P16R16 async mode, AudioIN EP (IF:4 EP:0x83) used for implicit feedback + ep = 0x83; + iface = usb_ifnum_to_if(dev, 4); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && After adding the patch, I verified that implicit feedback worked properly for playback, i.e. the OUT samples per packet tracked previous IN values. However, when attempting to perform simultaneous play and record via alsabat, alsaloop, or separate aplay/arecord, the system reports the 'endpoint already in use' error when attempting to set the format for the IN endpoint that is already in use (note that this dmesg log has customized printk messages): [Oct23 10:13] snd_usb_hw_params->set_format() [ +0.000014] set_format(): dev:b3856c00 alts:b67e8824 ifnum: 3 [ +0.001654] set_format()->usb_set_interface 3:1 [ +0.000006] Creating new playback data endpoint 03 [ +0.000004] Creating new capture data endpoint 83 <Begin Playback. SYNC endpoint also starts because feedback is implicit> [ +0.000027] snd_usb_pcm_prepare->set_format() [ +0.000005] set_format(): dev:b3856c00 alts:b67e8824 ifnum: 3 [ +0.000755] Setting params for EP 03 (type 0, 12 urbs), ret=0 [ +0.000017] Setting params for EP 83 (type 0, 12 urbs), ret=0 [ +0.000004] Starting DATA_EP 03 @b9224000 [ +0.000004] snd_usb_endpoint_implicit_feedback_sink (03) is TRUE [ +0.001495] Starting SYNC_EP 83 @b9222000 <After playing for 0.5s, start recording. Failure occurs because EP83 is already in use.> [ +0.493035] snd_usb_hw_params->set_format() [ +0.000016] set_format(): dev:b3856c00 alts:b67e88a4 ifnum: 4 [ +0.001549] set_format()->usb_set_interface 4:1 [ +0.000005] Re-using EP 83 in iface 4:1 [ +0.000028] snd_usb_pcm_prepare->set_format() [ +0.000004] set_format(): dev:b3856c00 alts:b67e88a4 ifnum: 4 *** [ +0.003196] *** snd_usb_endpoint_set_params(): EP #83: already in use [ +0.000003] configure_endpoint->endpoint_set_params: err=FFFFFFF0 I bought a used Fast Track Ultra device for comparison testing, and found that it behaved the same way as the CP2615 does with the implicit feedback quirk: in playback-only mode implicit feedback works properly, but play/record causes the 'EP already in use' error. Are there any suggestions on how I can resolve this issue? Any help would be greatly appreciated! Thanks, Bob Giaraffa

3 4

[alsa-devel] [PATCH v2 0/2] ALSA: usb-audio: fix PCM device order
by Takashi Iwai 30 Dec '19

30 Dec '19

Hi, this is a rewrite of the old USB-audio fix patch I obviously overlooked to adapt to 5.3 kernel with a preliminary cleanup patch. Totally untested, so any test / feedback would be appreciated. thanks, Takashi === Alexander Tsoy (1): ALSA: usb-audio: fix PCM device order Takashi Iwai (1): ALSA: usb-audio: Unify audioformat release code sound/usb/stream.c | 64 +++++++++++++++++++++++++++++++++++++++--------------- 1 file changed, 47 insertions(+), 17 deletions(-) -- 2.16.4

3 5

[alsa-devel] Surround speaker connection on Acer 8951G
by Sergey 'Jin' Bostandzhyan 28 Nov '19

28 Nov '19

Hi, I am not sure if this is the right place to ask or if I should just submit a bugreport, but you guys helped me out with a similar problem about 10 years ago, so I'll try here first. If I'm wrong here, then I apologize, just let me know. I have an Acer Aspire Ethos 8951G notebook which has 6 built in speakers for 5.1 surround sound, it's a Realtek ALC669X. The problem is that only two speakers are recognized by default. I tried fiddling around with hdajackretast and it does make a difference, but I somehow could not find a step-by-step approach on how to figure out what is what and how to map everything correctly, so some guidance would be very much appreciated. I even ended up in situations where I got channels supporting only mute which would influence muting of other channels, I really got lost there.. Here's my alsa-info output: http://alsa-project.org/db/?f=3adde87164f5fc349c3c78b211ee63e416ebf10b It would be great if we could figure it out and make it work by default. Basically that's what we did 10 years ago, back then I had an Acer Aspire 8920G with a ALC889 which had the same problem that got fixed by you and these days it just works out of the box :) Kind regards, Jin

3 22

[alsa-devel] Focusrite Scarlett 6i6 gen1 - input handling fix
by Jens Verwiebe 19 Nov '19

19 Nov '19

The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage. This patch introduces this functionality as to be seen in the mac or windows scarlett control. The correct address could already be found in the dump info, but was never used. Cheers ... Jens Signed-off-by: Jens Verwiebe <info(a)jensverwiebe.de> diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 7438e7c..f3bdac3 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -152,6 +152,7 @@ enum { SCARLETT_OUTPUTS, SCARLETT_SWITCH_IMPEDANCE, SCARLETT_SWITCH_PAD, + SCARLETT_SWITCH_GAIN, }; enum { @@ -202,6 +203,15 @@ struct scarlett_device_info { } }; +static const struct scarlett_mixer_elem_enum_info opt_gain = { + .start = 0, + .len = 2, + .offsets = {}, + .names = (char const * const []){ + "Lo", "Hi" + } +}; + static const struct scarlett_mixer_elem_enum_info opt_impedance = { .start = 0, .len = 2, @@ -664,8 +674,8 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, { .num = 1, .type = SCARLETT_SWITCH_PAD, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, }, .matrix_mux_init = { @@ -895,6 +905,15 @@ static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, if (err < 0) return err; break; + case SCARLETT_SWITCH_GAIN: + sprintf(mx, "Input %d Gain Switch", ctl->num); + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, + scarlett_ctl_enum_resume, 0x01, + 0x08, ctl->num, USB_MIXER_S16, 1, mx, + &opt_gain, &elem); + if (err < 0) + return err; + break; } } -- Jens Verwiebe Allerskehre 44 - 22309 Hamburg Tel.: +49 40 68 78 50 mobile: +49 172 400 49 07 mailto: info(a)jensverwiebe.de web: http://www.jensverwiebe.de

2 7

[alsa-devel] parameter for pulse device?
by frederik＠ofb.net 16 Nov '19

16 Nov '19

Dear ALSA, In my ~/.asoundrc, I have lines like this: pcm.mic { type pulse; device "mic" } pcm.monitor { type pulse; device "monitor"; } pcm.music { type pulse; device "music"; } to make it easier for ALSA-aware programs to input and output via PulseAudio, e.g.: ecasound -i alsa,mic -o alsa,monitor -etd:... However, I would like to simplify this and not have to update ~/.asoundrc every time I create a new PulseAudio device. Since ALSA has the ability for PCMs to take a parameter, I thought this might work with the "pulse" PCM and the PulseAudio device name. But I get an "error: Invalid argument" when trying to pass the device name as an argument to the "pulse" PCM: $ ecasound -o alsa,pulse:music -i some.wav ... ALSA lib conf.c:5014:(snd_config_expand) Unknown parameters music ALSA lib pcm.c:2564:(snd_pcm_open_noupdate) Unknown PCM pulse:music ERROR: Connecting chainsetup failed: "Enabling chainsetup: AUDIOIO-ALSA: ... Unable to open ALSA-device for playback; error: Invalid argument" Is there some magic with macros that I can use to accomplish this syntax, or can we add the ability for the "pulse" PCM to take a parameter naming the device? Thanks, Frederick

4 15

[alsa-devel] [PATCH v2] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint
by Yu-Hsuan Hsu 11 Nov '19

11 Nov '19

On KBL platform, the microphone is attached to external codec(rt5514) instead of PCH. However, TDM slot between PCH and codec is 16 bits only. In order to avoid setting wrong format, we should add a constraint to force to use 16 bits format forever. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- I have updated the commit message. Please see whether it is clear enough. Thanks. sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a01..67b276a65a8d2d 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -400,6 +400,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -- 2.23.0.351.gc4317032e6-goog

2 2