November 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Analog sound (ALC262) lost on linux (any distro and kernel) on a dual boot system
by Ploumistos Alexandros 03 Dec '14

03 Dec '14

Hello, A while back I had installed Ubuntu on a friend's laptop -a Sony Vaio VGN-FW21M- alongside windows Vista. Everything worked fine for a few years, until at some point last year linux went mute, while in windows both the analog and the digital outputs continued working. She couldn't remember doing something "out of the ordinary", but she was sure it was not after an update. Several months later, she upgraded to windows 7 but that didn't change anything. When I got my hands on the laptop, I checked for muted controls and other usual suspects, deprecated quirks (there were none) and all inputs and outputs of the system. HDMI audio works fine and so does the internal mic. Speakers, headphones and the external mic input do not work, even though pulse audio volume meter acts as if there were actually sounds being played. I followed all of Ubuntu's troubleshooting procedures, I removed and reinstalled alsa and pulse, I purged all configuration files, I created a new user and I even installed the latest alsa-packages from the "ALSA daily build snapshots" repository and nothing worked. I also tried booting into windows, setting all volume levels to the max and disabled power saving on all sound and multimedia devices. Next, I tried retasking the jacks one by one to no avail. I remembered that I had performed the installation with an Ubuntu 11.10 live CD, so I booted the computer off of that, but this time I got no sound. I tried with all the live CDs of a multitude of distros I had lying around, with kernels ranging from 2.6.35 to 3.16.6 and none of them worked, so I figured the issue must have something to do with windows. Is it possible that windows somehow started locking the audio chip on shutdown? If that is the case, can it be deduced from the logs and more importantly, can it be fixed? I am attaching the output of alsa-info and lspci as well as kernel logs. Thank you for your time.

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[alsa-devel] [PATCH 00/30 v3] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 03 Dec '14

03 Dec '14

This patchset is to enable ALSA Dice driver to handle more models, to improve ALSA firewire-speakers driver as OXFW driver. This patchset is for linux-next branch. == Updates from my previous post: [alsa-devel] [PATCH 00/29 v2] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/082844.ht… - Add a patch to fix some minor bugs (dice) - Improve handling config-rom strings for loud models (oxfw) - Fix a bug of mutual exclusive lock for AMDTP streams (oxfw) - Obsolete atomic_t for reference counters (dice/oxfw) Takashi Sakamoto (30): ALSA: dice: suppress checkpatch.pl warnings ALSA: dice: Rename structure and its members ALSA: dice: Move file to its own directory ALSA: dice: Split transaction functionality into a file ALSA: dice: Split stream functionality into a file ALSA: dice: Split PCM functionality into a file ALSA: dice: Split hwdep functionality into a file ALSA: dice: Split proc interface into a file ALSA: dice: Add new functions for constraints of PCM parameters ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: dice: remove experimental state ALSA: speakers: Rename to oxfw and rename some members ALSA: oxfw: Move to its own directory ALSA: oxfw: Split stream functionality to a new file and add a header file ALSA: oxfw: Split PCM functionality to a new file ALSA: oxfw: Split control functionality to a new file ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 26 +- sound/firewire/Makefile | 7 +- sound/firewire/dice.c | 1511 ---------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-hwdep.c | 190 ++++ sound/firewire/{ => dice}/dice-interface.h | 0 sound/firewire/dice/dice-midi.c | 194 ++++ sound/firewire/dice/dice-pcm.c | 424 ++++++++ sound/firewire/dice/dice-proc.c | 252 +++++ sound/firewire/dice/dice-stream.c | 407 ++++++++ sound/firewire/dice/dice-transaction.c | 382 +++++++ sound/firewire/dice/dice.c | 361 +++++++ sound/firewire/dice/dice.h | 189 ++++ sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw-command.c | 153 +++ sound/firewire/oxfw/oxfw-control.c | 283 ++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++ sound/firewire/oxfw/oxfw-midi.c | 207 ++++ sound/firewire/oxfw/oxfw-pcm.c | 425 ++++++++ sound/firewire/oxfw/oxfw-proc.c | 113 +++ sound/firewire/oxfw/oxfw-stream.c | 685 +++++++++++++ sound/firewire/oxfw/oxfw.c | 315 ++++++ sound/firewire/oxfw/oxfw.h | 145 +++ sound/firewire/speakers.c | 792 --------------- 26 files changed, 4944 insertions(+), 2319 deletions(-) delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-hwdep.c rename sound/firewire/{ => dice}/dice-interface.h (100%) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/dice/dice-pcm.c create mode 100644 sound/firewire/dice/dice-proc.c create mode 100644 sound/firewire/dice/dice-stream.c create mode 100644 sound/firewire/dice/dice-transaction.c create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw-command.c create mode 100644 sound/firewire/oxfw/oxfw-control.c create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c create mode 100644 sound/firewire/oxfw/oxfw-midi.c create mode 100644 sound/firewire/oxfw/oxfw-pcm.c create mode 100644 sound/firewire/oxfw/oxfw-proc.c create mode 100644 sound/firewire/oxfw/oxfw-stream.c create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h delete mode 100644 sound/firewire/speakers.c -- 2.1.0

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Re: [alsa-devel] ASoC: core: Call mute for cpu dais as well
by Dan Carpenter 02 Dec '14

02 Dec '14

Hello Ramesh Babu, This is a semi-automatic email about new static checker warnings. The patch ae11601b80b9: "ASoC: core: Call mute for cpu dais as well" from Oct 15, 2014, leads to the following Smatch complaint: sound/soc/soc-pcm.c:777 soc_pcm_prepare() error: we previously assumed 'cpu_dai->driver->ops' could be null (see line 755) sound/soc/soc-pcm.c 754 755 if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) { ^^^^^^^^^^^^^^^^^^^^ I don't think this can be NULL. Probably we should remove this check. 756 ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); 757 if (ret < 0) { 758 dev_err(cpu_dai->dev, "ASoC: DAI prepare error: %d\n", 759 ret); 760 goto out; 761 } 762 } 763 764 /* cancel any delayed stream shutdown that is pending */ 765 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && 766 rtd->pop_wait) { 767 rtd->pop_wait = 0; 768 cancel_delayed_work(&rtd->delayed_work); 769 } 770 771 snd_soc_dapm_stream_event(rtd, substream->stream, 772 SND_SOC_DAPM_STREAM_START); 773 774 for (i = 0; i < rtd->num_codecs; i++) 775 snd_soc_dai_digital_mute(rtd->codec_dais[i], 0, 776 substream->stream); 777 snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); ^^^^^^^ But if it were NULL then this newly introduced function call would Oops. 778 779 out: regards, dan carpenter

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[alsa-devel] [PATCH 2/2] ARM: tegra: Enable the mic-detect gpio on Acer Chromebook 13
by Dylan Reid 02 Dec '14

02 Dec '14

Enables the gpio-base mic detection on the Acer Chromebook 13. This gpio is set by the jack-detection chip when it notices either of the TRRS type headsets with a microphone. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- arch/arm/boot/dts/tegra124-nyan-big.dts | 2 ++ 1 file changed, 2 insertions(+) diff --git a/arch/arm/boot/dts/tegra124-nyan-big.dts b/arch/arm/boot/dts/tegra124-nyan-big.dts index 7d0784c..1c68ff9 100644 --- a/arch/arm/boot/dts/tegra124-nyan-big.dts +++ b/arch/arm/boot/dts/tegra124-nyan-big.dts @@ -1130,6 +1130,8 @@ clock-names = "pll_a", "pll_a_out0", "mclk"; nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(I, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-det-gpios = + <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; }; }; -- 2.1.2.330.g565301e

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[alsa-devel] [PATCH v7.1 00/19] Rework OMAP4+ HDMI audio support
by Jyri Sarha 02 Dec '14

02 Dec '14

The patches are based on: git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux.git for-next The base, the patches, and couple of additional not-to-be-merged omap2plus_defconfig patches can be found here: https://github.com/jsarha/linux.git omap-hdmi-audio It would make the most sense to get these in trough fbdev tree. So it would be nice to get acked-bys (if the patches are Ok) for ASoC side changes from appropriate maintainers. Changes since v7: - Squash: - "OMAPDSS: omapdss.h: Remove audio_state member of struct omap_dss_device" - into "OMAPDSS: Remove all references to obsolete HDMI audio callbacks" Changes since v6: - "ASoC: omap-hdmi-audio: Add platform device for OMAP HDMI audio support" - Fix Kconfig help text - Remove #include <sound/simple_card.h> - Use snd-soc-dummy codec instead of hdmi-audio-codec - Add: OMAPDSS: hdmi_wp: Protect reserved bits in hdmi_wp_audio_config_format() - Add: OMAPDSS: hdmi5_core: Initialize mandatory sample_order parameter - Add: "OMAPDSS: omapdss.h: Remove audio_state member of struct omap_dss_device" - "OMAPDSS: hdmi4: Register ASoC platform device for omap hdmi audio" - Register "omap-hdmi-audio" with PLATFORM_DEVID_AUTO - "OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio" - Register "omap-hdmi-audio" with PLATFORM_DEVID_AUTO - Add: "OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback" Jyri Sarha (19): OMAPDSS: hdmi_wp: Protect reserved bits in hdmi_wp_audio_config_format() OMAPDSS: hdmi5_core: Initialize mandatory sample_order parameter OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi: Remove most of OMAP[45]_DSS_HDMI_AUDIO ifdefs OMAPDSS: hdmi4_core: Remove unused hdmi4_audio_get_dma_port() OMAPDSS: hdmi_wp: Add function for getting audio dma address OMAPDSS: hdmi: Make hdmi structure public OMAPDSS: hdmi: Add pdev pointer for audio_pdev in HDMI DRV data ASoC: omap-hdmi-audio: Add platform device for OMAP HDMI audio support OMAPDSS: Kconfig: Remove HDMI audio booleans from Kconfig OMAPDSS: hdmi: Make hdmi_mode_has_audio() more user friedly OMAPDSS: hdmi.h: Add members to hdmi drvdata for audio implementation OMAPDSS: hdmi4: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi4: Register ASoC platform device for omap hdmi audio OMAPDSS: hdmi5: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: Remove all references to obsolete HDMI audio callbacks OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ----- .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 7 - drivers/video/fbdev/omap2/dss/hdmi.h | 38 +- drivers/video/fbdev/omap2/dss/hdmi4.c | 269 +++++++------ drivers/video/fbdev/omap2/dss/hdmi4_core.c | 14 - drivers/video/fbdev/omap2/dss/hdmi4_core.h | 4 - drivers/video/fbdev/omap2/dss/hdmi5.c | 265 ++++++------- drivers/video/fbdev/omap2/dss/hdmi5_core.c | 9 +- drivers/video/fbdev/omap2/dss/hdmi5_core.h | 2 - drivers/video/fbdev/omap2/dss/hdmi_common.c | 2 - drivers/video/fbdev/omap2/dss/hdmi_wp.c | 16 +- include/sound/omap-hdmi-audio.h | 43 +++ include/video/omapdss.h | 40 -- sound/soc/omap/Kconfig | 26 +- sound/soc/omap/Makefile | 6 +- sound/soc/omap/omap-hdmi-audio.c | 407 ++++++++++++++++++++ sound/soc/omap/omap-hdmi-card.c | 87 ----- sound/soc/omap/omap-hdmi.c | 364 ----------------- sound/soc/omap/omap-hdmi.h | 38 -- 20 files changed, 781 insertions(+), 1011 deletions(-) create mode 100644 include/sound/omap-hdmi-audio.h create mode 100644 sound/soc/omap/omap-hdmi-audio.c delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: Intel: Move capture PCM pin to PCM0 for Broadwell/Haswell
by Jie Yang 02 Dec '14

02 Dec '14

Move capture PCM pin from PCM4 to PCM0 for Broadwell/Haswell. This will allow us to integrate with pulseaudio better for usually default device is set to 0. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/broadwell.c | 15 ++------------- sound/soc/intel/haswell.c | 14 ++------------ sound/soc/intel/sst-haswell-pcm.c | 34 ++++++++++++++++------------------ 3 files changed, 20 insertions(+), 43 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 52cb764..c256764 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -171,7 +171,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { /* Front End DAI links */ { .name = "System PCM", - .stream_name = "System Playback", + .stream_name = "System Playback/Capture", .cpu_dai_name = "System Pin", .platform_name = "haswell-pcm-audio", .dynamic = 1, @@ -180,6 +180,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .init = broadwell_rtd_init, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, + .dpcm_capture = 1, }, { .name = "Offload0", @@ -214,18 +215,6 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, - { - .name = "Capture PCM", - .stream_name = "Capture", - .cpu_dai_name = "Capture Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, - /* Back End DAI links */ { /* SSP0 - Codec */ diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 3981982..cb8a482 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -109,7 +109,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { /* Front End DAI links */ { .name = "System", - .stream_name = "System Playback", + .stream_name = "System Playback/Capture", .cpu_dai_name = "System Pin", .platform_name = "haswell-pcm-audio", .dynamic = 1, @@ -118,6 +118,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .init = haswell_rtd_init, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, + .dpcm_capture = 1, }, { .name = "Offload0", @@ -152,17 +153,6 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, - { - .name = "Capture", - .stream_name = "Capture", - .cpu_dai_name = "Capture Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, /* Back End DAI links */ { diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 5f70d81..0180b38 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -309,7 +309,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ - SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; @@ -396,8 +396,14 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, /* DSP stream type depends on DAI ID */ switch (rtd->cpu_dai->id) { case 0: - stream_type = SST_HSW_STREAM_TYPE_SYSTEM; - module_id = SST_HSW_MODULE_PCM_SYSTEM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + stream_type = SST_HSW_STREAM_TYPE_SYSTEM; + module_id = SST_HSW_MODULE_PCM_SYSTEM; + } + else { + stream_type = SST_HSW_STREAM_TYPE_CAPTURE; + module_id = SST_HSW_MODULE_PCM_CAPTURE; + } break; case 1: case 2: @@ -410,10 +416,6 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, path_id = SST_HSW_STREAM_PATH_SSP0_OUT; module_id = SST_HSW_MODULE_PCM_REFERENCE; break; - case 4: - stream_type = SST_HSW_STREAM_TYPE_CAPTURE; - module_id = SST_HSW_MODULE_PCM_CAPTURE; - break; default: dev_err(rtd->dev, "error: invalid DAI ID %d\n", rtd->cpu_dai->id); @@ -779,6 +781,13 @@ static struct snd_soc_dai_driver hsw_dais[] = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Analog Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, + }, }, { /* PCM */ @@ -815,17 +824,6 @@ static struct snd_soc_dai_driver hsw_dais[] = { .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, - { - .name = "Capture Pin", - .id = HSW_PCM_DAI_ID_CAPTURE, - .capture = { - .stream_name = "Analog Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, - }, - }, }; static const struct snd_soc_dapm_widget widgets[] = { -- 1.9.1

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[alsa-devel] [PATCH] ALSA: usb: Add support for Zoom R16/24 capture and midi interfaces
by Panu Matilainen 01 Dec '14

01 Dec '14

This makes the midi interface and capture work out of the box with R16 (and presumably R24 too but untested). Playback stream would also seem to function fine except for one caveat: no sound is produced, so it is disabled for now. Signed-off-by: Panu Matilainen <pmatilai(a)laiskiainen.org> --- sound/usb/quirks-table.h | 34 ++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 8 ++++++++ 2 files changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752..73621c9 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3202,6 +3202,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { + /* ZOOM R16/24 in audio interface mode */ + USB_DEVICE(0x1686, 0x00dd), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + /* Control */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Playback, sound is missing atm */ + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Capture */ + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + /* Midi */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + }, + } + } +}, + +{ /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a5941f8..2e49193 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1160,6 +1160,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, break; } } + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && + (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(1); } /* -- 1.9.3

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[alsa-devel] [PATCH v2] ALSA: usb: Add support for Zoom R16/24 capture and midi interfaces
by Panu Matilainen 01 Dec '14

01 Dec '14

This makes the midi interface and capture work out of the box with R16 (and presumably R24 too but untested). Playback stream would also seem to function fine except for one caveat: no sound is produced, so it is disabled for now. Mixer descriptors are garbage and will require further quirks to enable functionality, also disabled here. Signed-off-by: Panu Matilainen <pmatilai(a)laiskiainen.org> --- sound/usb/quirks-table.h | 40 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 8 ++++++++ 2 files changed, 48 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752..c955663 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3203,6 +3203,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* + * ZOOM R16/24 in audio interface mode. + * Mixer descriptors are garbage, further quirks will be needed + * to make any of it functional, thus disabled for now. + * Playback stream appears to start and run fine but no sound + * is produced, so also disabled for now. + */ + USB_DEVICE(0x1686, 0x00dd), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + /* Mixer */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Playback */ + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Capture */ + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + /* Midi */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + }, + } + } +}, + +{ + /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a5941f8..2e49193 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1160,6 +1160,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, break; } } + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && + (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(1); } /* -- 1.9.3

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[alsa-devel] [GIT PULL] Add sst audio firmware for baytrail platforms
by Vinod Koul 01 Dec '14

01 Dec '14

Hello Linux firmware maintainers, Please PULL to receive the Intel audio firmware for audio DSP in Baytrail platforms. This firmware is using the same license as another version of the Baytrail firmware already available in linux-firmware git. The following changes since commit 0e5f63771d0df6d7859f7c4100a74d737c62ac88: are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git byt Vinod Koul (1): linux-firmware: add sst audio firmware for baytrail platforms WHENCE | 8 ++++++++ intel/fw_sst_0f28.bin | Bin 0 -> 647078 bytes 2 files changed, 8 insertions(+), 0 deletions(-) create mode 100644 intel/fw_sst_0f28.bin Thanks -- ~Vinod

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[alsa-devel] [PATCH] Sound: drivers: virmidi: fixed code style issues
by Kyle Chamberlin 30 Nov '14

30 Nov '14

Fixed some minor code style issues and also removed some assignments inside of if conditionals. Signed-off-by: Kyle Chamberlin <KyleChamberlin(a)project20million.org> --- sound/drivers/virmidi.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index b17872429..0af88d9 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -99,30 +99,33 @@ static int snd_virmidi_probe(struct platform_device *devptr) if (midi_devs[dev] > MAX_MIDI_DEVICES) { snd_printk(KERN_WARNING - "too much midi devices for virmidi %d: " - "force to use %d\n", dev, MAX_MIDI_DEVICES); + "too much midi devices for virmidi %d: force to use %d\n", + dev, MAX_MIDI_DEVICES); midi_devs[dev] = MAX_MIDI_DEVICES; } for (idx = 0; idx < midi_devs[dev]; idx++) { struct snd_rawmidi *rmidi; struct snd_virmidi_dev *rdev; - if ((err = snd_virmidi_new(card, idx, &rmidi)) < 0) + + err = snd_virmidi_new(card, idx, &rmidi); + if (err < 0) goto __nodev; rdev = rmidi->private_data; vmidi->midi[idx] = rmidi; strcpy(rmidi->name, "Virtual Raw MIDI"); rdev->seq_mode = SNDRV_VIRMIDI_SEQ_DISPATCH; } - + strcpy(card->driver, "VirMIDI"); strcpy(card->shortname, "VirMIDI"); sprintf(card->longname, "Virtual MIDI Card %i", dev + 1); - if ((err = snd_card_register(card)) == 0) { + err = snd_card_register(card); + if (err) { platform_set_drvdata(devptr, card); return 0; } - __nodev: +__nodev: snd_card_free(card); return err; } @@ -157,13 +160,15 @@ static int __init alsa_card_virmidi_init(void) { int i, cards, err; - if ((err = platform_driver_register(&snd_virmidi_driver)) < 0) + err = platform_driver_register(&snd_virmidi_driver); + if (err < 0) return err; cards = 0; for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; - if (! enable[i]) + + if (!enable[i]) continue; device = platform_device_register_simple(SND_VIRMIDI_DRIVER, i, NULL, 0); -- 2.1.3

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