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

[alsa-devel] tegra30 tdm audio support
by Ben Dooks 20 Mar '20

20 Mar '20

TDM audio support for tegra30 devices. v2: - moved edge-control and data-offset to the set_fmt callbacks - fixed dev_dbg in set_tdm callback - fixed dev_dbg message contents in set_tdm callback - changed fsync width to be permanently 1 clock v3: - cleanup fsync patch - fix rebase issue with tdm patch v4: - fix comment style issues - change tdm-a to data-offset 1 v5: - fix format on tdm-b

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[alsa-devel] Add headset Mic no shutup for ALC283
by Kailang 26 Feb '20

26 Feb '20

Hi Takashi, ChromeBox machine had humming noise from external skeaker plugin at codec D3 state. BR, Kailang

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[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

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[alsa-devel] [PATCH] ASoC: Intel: mrfld: fix incorrect check on p->sink
by Colin King 25 Feb '20

25 Feb '20

From: Colin Ian King <colin.king(a)canonical.com> The check on p->sink looks bogus, I believe it should be p->source since the following code blocks are related to p->source. Fix this by replacing p->sink with p->source. Addresses-Coverity: ("Copy-paste error") Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- [ Note: this has not been tested ] --- sound/soc/intel/atom/sst-atom-controls.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f1..f883c9340eee 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1333,7 +1333,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) dai->capture_widget->name); w = dai->capture_widget; snd_soc_dapm_widget_for_each_source_path(w, p) { - if (p->connected && !p->connected(w, p->sink)) + if (p->connected && !p->connected(w, p->source)) continue; if (p->connect && p->source->power && -- 2.24.0

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[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

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[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.

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[alsa-devel] [PATCH] ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen
by Jaroslav Kysela 11 Feb '20

11 Feb '20

The auto-parser assigns the bass speaker to DAC3 (NID 0x06) which is without the volume control. I do not see a reason to use DAC2, because the shared output to all speakers produces the sufficient and well balanced sound. The stereo support is enough for this purpose (laptop). Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d2bf70a1d2fd..a596790d4245 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5544,6 +5544,16 @@ static void alc295_fixup_disable_dac3(struct hda_codec *codec, } } +/* force NID 0x17 (Bass Speaker) to DAC1 to share it with the main speaker */ +static void alc285_fixup_speaker2_to_dac1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + hda_nid_t conn[1] = { 0x02 }; + snd_hda_override_conn_list(codec, 0x17, 1, conn); + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -5846,6 +5856,7 @@ enum { ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, + ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -6646,6 +6657,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_disable_dac3, }, + [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7221,6 +7236,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), @@ -7405,6 +7421,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, + {.id = ALC285_FIXUP_SPEAKER2_TO_DAC1, .name = "alc285-speaker2-to-dac1"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, {.id = ALC298_FIXUP_SPK_VOLUME, .name = "alc298-spk-volume"}, -- 2.20.1

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[alsa-devel] [PATCH v2 0/2] Update FE/BE trigger order based on the command
by Ranjani Sridharan 09 Feb '20

09 Feb '20

Currently, the trigger orders SND_SOC_DPCM_TRIGGER_PRE/POST determine the order in which FE DAI and BE DAI are triggered. In the case of SND_SOC_DPCM_TRIGGER_PRE, the FE DAI is triggered before the BE DAI and in the case of SND_SOC_DPCM_TRIGGER_POST, the BE DAI is triggered before the FE DAI. And this order remains the same irrespective of the trigger command. In the case of the SOF driver, during playback, the FW expects the BE DAI to be triggered before the FE DAI during the START trigger. The BE DAI trigger handles the starting of Link DMA and so it must be started before the FE DAI is started to prevent xruns during pause/release. This can be addressed by setting the trigger order for the FE dai link to SND_SOC_DPCM_TRIGGER_POST. But during the STOP trigger, the FW expects the FE DAI to be triggered before the BE DAI. Retaining the same order during the START and STOP commands, results in FW error as the DAI component in the FW is still active. The issue can be fixed by mirroring the trigger order of FE and BE DAI's during the START and STOP trigger. So, with the trigger order set to SND_SOC_DPCM_TRIGGER_PRE, the FE DAI will be trigger first during SNDRV_PCM_TRIGGER_START/STOP/RESUME and the BE DAI will be triggered first during the STOP/SUSPEND/PAUSE commands. Conversely, with the trigger order set to SND_SOC_DPCM_TRIGGER_POST, the BE DAI will be triggered first during the SNDRV_PCM_TRIGGER_START/STOP/RESUME commands and the FE DAI will be triggered first during the SNDRV_PCM_TRIGGER_STOP/SUSPEND/PAUSE commands. The above change has been tested with the Skylake driver on the Kabylake based chromebook and legacy HDA driver on Kabylake based Dell XPS13. It has also been tested on all platforms with SOF CI without any regressions for the past 3 weeks. More details on the issue can be found here: https://github.com/thesofproject/linux/issues/1160 Changes in v2: Fixed comments on function name and consolidating the switch cases. Ranjani Sridharan (2): ASoC: pcm: update FE/BE trigger order based on the command ASoC: SOF: topology: set trigger order for FE DAI link sound/soc/soc-pcm.c | 95 ++++++++++++++++++++++++++++------------ sound/soc/sof/topology.c | 4 ++ 2 files changed, 72 insertions(+), 27 deletions(-) -- 2.17.1

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[alsa-devel] USB Audio Interface / Denon MC7000 and MC8000 controller
by Tobias 07 Feb '20

07 Feb '20

Hello dear ALSA developers. I have purchased a MC7000 controller in order to control MIXXX on Ubuntu 16.04. According to the Denon specification the controller should have been class compliant but there is an issue with the Audio interface to work properly giving following message ... $ dmesg ... |[ 74.522831] usb 1-1.3: new high-speed USB device number 6 using xhci_hcd [ 74.623784] usb 1-1.3: New USB device found, idVendor=15e4, idProduct=8004 [ 74.623789] usb 1-1.3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 74.623793] usb 1-1.3: Product: DENON DJ MC7000 [ 74.623796] usb 1-1.3: Manufacturer: DENON DJ [ 74.623798] usb 1-1.3: SerialNumber: 201603 [ 74.625134] usb 1-1.3: clock source 65 is not valid, cannot use "|||clock source 65 is not valid, cannot use" is repeated uncountable times then |I found a discussion here for the MC8000 Audio interface which was not working and showing the same error message: https://alsa-user.narkive.com/2tDAO87f/troubleshooting-new-usb-audio-device… The solution was to change sound/usb/clock.c and recompile the kernel. This is not practical for daily use case so I would like to ask if the ALSA team could possibly find a permanent fix for ordinary users. To get my outputs for aplay -l aplay -L |||aplay -D plughw:CARD=MC7000,DEV=0 Grimmaldika-MakeBelieve.wav |lsusb -v mixxx --controllerDebug|| JackServer error message| you may have a look at the Mixxx Community Forums here(just on page, you don't need to find and read on multiple pages) https://www.mixxx.org/forums/viewtopic.php?f=7&t=12962&start=10 If you need any more information then please let me know. Thanks a lot for your support on this matter. Cheers! OsZ ||

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[alsa-devel] [PATCH v5 00/17] soundwire: intel: implement new ASoC interfaces
by Pierre-Louis Bossart 03 Feb '20

03 Feb '20

This patchset applies on top of soundwire/next, now that the interface definitions are merged. The changes are essentially a removal of the platform devices, replaced by explicit device(s)/driver for the SoundWire Master(s), and the implementation of the new interfaces required to scan the ACPI tables, probe the links and start them. The missing prepare, trigger and setup ASoC callbacks are also implemented. The hw_params and free callbacks use the new interfaces as well. While there are quite a few lines of code changed, this is mostly about interface changes. The next series will contain more functional changes and deal with race conditions on probe, enumeration and suspend/resume issues. Thanks to Vinod Kould and GregKH for the reviews on previous versions, much appreciated. Changes since v4: (feedback from GregKH except last point flagged by Intel validation) Clarified error handling with uevents Added better commit messages and rationale behind exposing a Master Device. Used 'master_device' instead of 'md' when possible (kept the shortcut with really long function names) Added namespace check support for Intel code (the Cadence and core stuff should be done in a separate patchset). Fixed GPLv2 license/EXPORT_SYMBOL_GPL error for the Master Device code. Fixed missing error handling in sdw_md_add Added kerneldoc comments for sdw_md_driver structure (with explanation on what 'md' stands for) Fixed NULL pointer assignment leading to issues with driver_unregister Changes since v3: One line change to re-add EXPORT_SYMBOL Add missing driver_registration Changes since v2: moved uevent handling to slave_type (Vinod) Changes since v1: fix typo (Vinod) removed uevent open for Master (Vinod) clarified commit messages (Cezary) no functionality change Bard Liao (1): soundwire: register master device driver Pierre-Louis Bossart (13): soundwire: renames to prepare support for master drivers/devices soundwire: rename dev_to_sdw_dev macro soundwire: rename drv_to_sdw_slave_driver macro soundwire: bus_type: rename sdw_drv_ to sdw_slave_drv soundwire: intel: rename res field as link_res soundwire: add support for sdw_slave_type soundwire: slave: move uevent handling to slave device level soundwire: add initial definitions for sdw_master_device soundwire: intel: remove platform devices and use 'Master Devices' instead soundwire: intel: free all resources on hw_free() soundwire: intel_init: add implementation of sdw_intel_enable_irq() soundwire: intel_init: use EXPORT_SYMBOL_NS soundwire: intel: use EXPORT_SYMBOL_NS Rander Wang (3): soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback drivers/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus.h | 2 + drivers/soundwire/bus_type.c | 70 ++++--- drivers/soundwire/intel.c | 281 +++++++++++++++++++++----- drivers/soundwire/intel.h | 8 +- drivers/soundwire/intel_init.c | 312 ++++++++++++++++++++++------- drivers/soundwire/master.c | 64 ++++++ drivers/soundwire/slave.c | 10 +- include/linux/soundwire/sdw.h | 42 +++- include/linux/soundwire/sdw_type.h | 34 +++- 12 files changed, 668 insertions(+), 163 deletions(-) create mode 100644 drivers/soundwire/master.c -- 2.20.1

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