January 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "IIO: ADC: stm32-dfsdm: fix static check warning" to the asoc tree
by Mark Brown 16 Jan '18

16 Jan '18

The patch IIO: ADC: stm32-dfsdm: fix static check warning has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1175d0f9f4031ce02845f6f843f58a9caaee7817 Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Date: Mon, 15 Jan 2018 10:00:26 +0100 Subject: [PATCH] IIO: ADC: stm32-dfsdm: fix static check warning iio_priv does not return an error pointer, so check is not valid. Patch suppresses it. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/iio/adc/stm32-dfsdm-adc.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/drivers/iio/adc/stm32-dfsdm-adc.c b/drivers/iio/adc/stm32-dfsdm-adc.c index 3fe9b34ac6af..daa026d6a94f 100644 --- a/drivers/iio/adc/stm32-dfsdm-adc.c +++ b/drivers/iio/adc/stm32-dfsdm-adc.c @@ -1099,10 +1099,6 @@ static int stm32_dfsdm_adc_probe(struct platform_device *pdev) } adc = iio_priv(iio); - if (IS_ERR(adc)) { - dev_err(dev, "%s: Failed to allocate ADC\n", __func__); - return PTR_ERR(adc); - } adc->dfsdm = dev_get_drvdata(dev->parent); iio->dev.parent = dev; -- 2.15.1

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[alsa-devel] ASoC: fix regression by adding back .read/.write
by Kuninori Morimoto 16 Jan '18

16 Jan '18

Hi Mark As Tony mentioned in "PM regression in next" mail thread, below drivers need .read/.write back on codec driver. twl4030/twl6040 side will be fixed by Tony's patch, thus, I post for remains. sound/soc/codecs/cx20442.c sound/soc/codecs/tlv320dac33.c sound/soc/codecs/twl4030.c sound/soc/codecs/twl6040.c sound/soc/codecs/uda1380.c Because of this, my posted codec/platform conversion patch-set will have conflict on above drivers. I'm happy to post v2 patch for these when you accepted that series. This .read/.write callback is implemented in Codec driver side only. We need it in Component side too. Last patch is for it. Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH v2] ALSA: pcm: Fix trailing semicolon
by Luis de Bethencourt 16 Jan '18

16 Jan '18

The trailing semicolon is an empty statement that does no operation. Removing it since it doesn't do anything. Signed-off-by: Luis de Bethencourt <luisbg(a)kernel.org> --- Changes from v1: - Fix commit message sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 3f13e8da0812..51104df924e1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3446,7 +3446,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap); int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { - struct snd_pcm_runtime *runtime = substream->runtime;; + struct snd_pcm_runtime *runtime = substream->runtime; area->vm_page_prot = pgprot_noncached(area->vm_page_prot); return vm_iomap_memory(area, runtime->dma_addr, runtime->dma_bytes); -- 2.15.1

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[alsa-devel] [PATCH] ALSA: pcm: Fix trailing semicolon
by Luis de Bethencourt 16 Jan '18

16 Jan '18

From: Ubuntu <luisbg(a)kernel.org> The trailing semicolon is an empty statement that does no operation. Removing it since it doesn't do anything. Signed-off-by: Luis de Bethencourt <luisbg(a)kernel.org> --- Hi, After fixing the same thing in drivers/staging/rtl8723bs/, Joe Perches suggested I fix it treewide [0]. Best regards Luis [0] http://driverdev.linuxdriverproject.org/pipermail/driverdev-devel/2018-Janu… [1] http://driverdev.linuxdriverproject.org/pipermail/driverdev-devel/2018-Janu… sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 3f13e8da0812..51104df924e1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3446,7 +3446,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap); int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { - struct snd_pcm_runtime *runtime = substream->runtime;; + struct snd_pcm_runtime *runtime = substream->runtime; area->vm_page_prot = pgprot_noncached(area->vm_page_prot); return vm_iomap_memory(area, runtime->dma_addr, runtime->dma_bytes); -- 2.15.1

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[alsa-devel] Sound Blaster Z - Sound from front case port, but not back
by Frederique 16 Jan '18

16 Jan '18

I am using a Sound Blaster Z on Debian. The back microphone port works as expected. I can only manage to get audio out of the card by connecting the case audio header to the sound card, and then using the case front audio port and having the headphone port option enabled in alsamixer. It is an inconvenience as now I have to use an extension cable to route things around (microphone cable at the back, audio cable to the front), having a headset. Does anyone know a solution to this? How hard can it be to make the back audio part do the same thing as the front audio port?

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[alsa-devel] [PATCH v7 00/17] Add support for Cirrus Logic CS47L35/L85/L90/L91 codecs
by Richard Fitzgerald 16 Jan '18

16 Jan '18

IMPORTANT: These patches have to be submitted as an atomic block because of build interdependencies so you'll need to agree who is going to take them all through a branch. The Cirrus Logic CS47L35, CS47L85, CS47L90/91 codecs are complex audio SoC devices. In addition to the core audio capability they have onboard GPIO, regulators, DSPs and interrupt controller and a large register map space accessed over SPI or I2C. This family of codecs is based around common IP blocks and they are managed by a set of common drivers referred to as "Madera". Patches still waiting for review/ack: #7-8 regulator #9 irqchip #14-17 ASoC Richard Fitzgerald (17): mfd: madera: Add register definitions for Cirrus Logic Madera codecs mfd: madera: Add DT bindings for Cirrus Logic Madera codecs mfd: madera: Add common support for Cirrus Logic Madera codecs mfd: madera: Register map tables for Cirrus Logic CS47L35 mfd: madera: Register map tables for Cirrus Logic CS47L85 mfd: madera: Register map tables for Cirrus Logic CS47L90/91 regulator: arizona-micsupp: Add support for Cirrus Logic Madera codecs regulator: arizona-ldo1: Add support for Cirrus Logic Madera codecs irqchip: Add driver for Cirrus Logic Madera codecs pinctrl: madera: Add DT bindings for Cirrus Logic Madera codecs pinctrl: madera: Add driver for Cirrus Logic Madera codecs gpio: madera: Support Cirrus Logic Madera class codecs ASoC: madera: Add DT bindings for Cirrus Logic Madera codecs ASoC: madera: Add common support for Cirrus Logic Madera codecs ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35 ASoC: cs47l85: Add codec driver for Cirrus Logic CS47L85 ASoC: cs47l90: Add codec driver for Cirrus Logic CS47L90 Documentation/devicetree/bindings/mfd/madera.txt | 102 + .../bindings/pinctrl/cirrus,madera-pinctrl.txt | 99 + .../bindings/regulator/arizona-regulator.txt | 3 +- Documentation/devicetree/bindings/sound/madera.txt | 67 + MAINTAINERS | 23 + drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-madera.c | 196 + drivers/irqchip/Kconfig | 3 + drivers/irqchip/Makefile | 1 + drivers/irqchip/irq-madera.c | 301 ++ drivers/mfd/Kconfig | 48 + drivers/mfd/Makefile | 13 + drivers/mfd/cs47l35-tables.c | 1610 +++++++ drivers/mfd/cs47l85-tables.c | 3010 +++++++++++++ drivers/mfd/cs47l90-tables.c | 2675 ++++++++++++ drivers/mfd/madera-core.c | 599 +++ drivers/mfd/madera-i2c.c | 140 + drivers/mfd/madera-spi.c | 139 + drivers/mfd/madera.h | 44 + drivers/pinctrl/Kconfig | 1 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/cirrus/Kconfig | 15 + drivers/pinctrl/cirrus/Makefile | 11 + drivers/pinctrl/cirrus/pinctrl-cs47l35.c | 46 + drivers/pinctrl/cirrus/pinctrl-cs47l85.c | 60 + drivers/pinctrl/cirrus/pinctrl-cs47l90.c | 58 + drivers/pinctrl/cirrus/pinctrl-madera.c | 1074 +++++ drivers/pinctrl/cirrus/pinctrl-madera.h | 40 + drivers/regulator/Kconfig | 15 +- drivers/regulator/arizona-ldo1.c | 82 +- drivers/regulator/arizona-micsupp.c | 71 +- include/dt-bindings/sound/madera.h | 27 + include/linux/irqchip/irq-madera-pdata.h | 23 + include/linux/irqchip/irq-madera.h | 98 + include/linux/mfd/madera/core.h | 196 + include/linux/mfd/madera/pdata.h | 61 + include/linux/mfd/madera/registers.h | 3916 +++++++++++++++++ include/sound/madera-pdata.h | 61 + sound/soc/codecs/Kconfig | 23 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/cs47l35.c | 1671 ++++++++ sound/soc/codecs/cs47l85.c | 2677 ++++++++++++ sound/soc/codecs/cs47l90.c | 2651 ++++++++++++ sound/soc/codecs/madera.c | 4439 ++++++++++++++++++++ sound/soc/codecs/madera.h | 470 +++ 46 files changed, 26865 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/madera.txt create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 drivers/gpio/gpio-madera.c create mode 100644 drivers/irqchip/irq-madera.c create mode 100644 drivers/mfd/cs47l35-tables.c create mode 100644 drivers/mfd/cs47l85-tables.c create mode 100644 drivers/mfd/cs47l90-tables.c create mode 100644 drivers/mfd/madera-core.c create mode 100644 drivers/mfd/madera-i2c.c create mode 100644 drivers/mfd/madera-spi.c create mode 100644 drivers/mfd/madera.h create mode 100644 drivers/pinctrl/cirrus/Kconfig create mode 100644 drivers/pinctrl/cirrus/Makefile create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l35.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l85.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l90.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.h create mode 100644 include/dt-bindings/sound/madera.h create mode 100644 include/linux/irqchip/irq-madera-pdata.h create mode 100644 include/linux/irqchip/irq-madera.h create mode 100644 include/linux/mfd/madera/core.h create mode 100644 include/linux/mfd/madera/pdata.h create mode 100644 include/linux/mfd/madera/registers.h create mode 100644 include/sound/madera-pdata.h create mode 100644 sound/soc/codecs/cs47l35.c create mode 100644 sound/soc/codecs/cs47l85.c create mode 100644 sound/soc/codecs/cs47l90.c create mode 100644 sound/soc/codecs/madera.c create mode 100644 sound/soc/codecs/madera.h -- 2.11.0

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[alsa-devel] [PATCH] sound/pci/hda: depends on instead of select for INPUT
by Randy Dunlap 15 Jan '18

15 Jan '18

From: Randy Dunlap <rdunlap(a)infradead.org> Drivers should not 'select' a subsystem. Instead they should depend on it. If the subsystem is disabled, the user probably did that for a purpose and one driver shouldn't be changing that. This also makes all sound/ drivers consistent w.r.t depending on INPUT instead of selecting it. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org (moderated for non-subscribers) --- sound/pci/hda/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- lnx-415-rc8.orig/sound/pci/hda/Kconfig +++ lnx-415-rc8/sound/pci/hda/Kconfig @@ -87,8 +87,8 @@ config SND_HDA_PATCH_LOADER config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" + depends on INPUT select SND_HDA_GENERIC - select INPUT help Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880.

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[alsa-devel] [PATCH] ALSA: seq: Process queue tempo/ppq change in a shot
by Takashi Iwai 15 Jan '18

15 Jan '18

The SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO ioctl sets the tempo and the ppq in a single call, while the current implementation updates each value one by one. This is a bit racy, and also suboptimal from the performance POV, as each call does re-acquire the lock and invokes the update of ALSA timer resolution. This patch reorganizes the code slightly so that we change both the tempo and the ppq in a shot. The skew value can be put into the same lock, but this is rather a rarely used feature and completely independent from the temp/ppq (it's evaluated only in the interrupt), so it's left as it was. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_queue.c | 4 +--- sound/core/seq/seq_timer.c | 13 ++++++++----- sound/core/seq/seq_timer.h | 2 +- 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 79e0c5604ef8..0428e9061b47 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -497,9 +497,7 @@ int snd_seq_queue_timer_set_tempo(int queueid, int client, return -EPERM; } - result = snd_seq_timer_set_tempo(q->timer, info->tempo); - if (result >= 0) - result = snd_seq_timer_set_ppq(q->timer, info->ppq); + result = snd_seq_timer_set_tempo_ppq(q->timer, info->tempo, info->ppq); if (result >= 0 && info->skew_base > 0) result = snd_seq_timer_set_skew(q->timer, info->skew_value, info->skew_base); diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index b80985fbc334..23167578231f 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -191,14 +191,15 @@ int snd_seq_timer_set_tempo(struct snd_seq_timer * tmr, int tempo) return 0; } -/* set current ppq */ -int snd_seq_timer_set_ppq(struct snd_seq_timer * tmr, int ppq) +/* set current tempo and ppq in a shot */ +int snd_seq_timer_set_tempo_ppq(struct snd_seq_timer *tmr, int tempo, int ppq) { + int changed; unsigned long flags; if (snd_BUG_ON(!tmr)) return -EINVAL; - if (ppq <= 0) + if (tempo <= 0 || ppq <= 0) return -EINVAL; spin_lock_irqsave(&tmr->lock, flags); if (tmr->running && (ppq != tmr->ppq)) { @@ -208,9 +209,11 @@ int snd_seq_timer_set_ppq(struct snd_seq_timer * tmr, int ppq) pr_debug("ALSA: seq: cannot change ppq of a running timer\n"); return -EBUSY; } - + changed = (tempo != tmr->tempo) || (ppq != tmr->ppq); + tmr->tempo = tempo; tmr->ppq = ppq; - snd_seq_timer_set_tick_resolution(tmr); + if (changed) + snd_seq_timer_set_tick_resolution(tmr); spin_unlock_irqrestore(&tmr->lock, flags); return 0; } diff --git a/sound/core/seq/seq_timer.h b/sound/core/seq/seq_timer.h index 9506b661fe5b..62f390671096 100644 --- a/sound/core/seq/seq_timer.h +++ b/sound/core/seq/seq_timer.h @@ -131,7 +131,7 @@ int snd_seq_timer_stop(struct snd_seq_timer *tmr); int snd_seq_timer_start(struct snd_seq_timer *tmr); int snd_seq_timer_continue(struct snd_seq_timer *tmr); int snd_seq_timer_set_tempo(struct snd_seq_timer *tmr, int tempo); -int snd_seq_timer_set_ppq(struct snd_seq_timer *tmr, int ppq); +int snd_seq_timer_set_tempo_ppq(struct snd_seq_timer *tmr, int tempo, int ppq); int snd_seq_timer_set_position_tick(struct snd_seq_timer *tmr, snd_seq_tick_time_t position); int snd_seq_timer_set_position_time(struct snd_seq_timer *tmr, snd_seq_real_time_t position); int snd_seq_timer_set_skew(struct snd_seq_timer *tmr, unsigned int skew, unsigned int base); -- 2.15.1

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[alsa-devel] [PATCH 1/4] ASoC: codecs: tas5720: add basic support for TAS5722 devices
by Andrew F. Davis 15 Jan '18

15 Jan '18

From: Andreas Dannenberg <dannenberg(a)ti.com> The TI TAS5722 digital amplifier is very similar to the TAS5720 from an overall and register map perspective. Therefore the existing driver can be extended easily to support this additional device. This commit allows TAS5722 devices to be used in a "subset" type of fashion, without exposing any of the additional features they offer. Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- .../devicetree/bindings/sound/tas5720.txt | 4 ++- sound/soc/codecs/tas5720.c | 38 ++++++++++++++++++---- sound/soc/codecs/tas5720.h | 1 + 3 files changed, 35 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt index 40d94f82beb3..7481653fe8e3 100644 --- a/Documentation/devicetree/bindings/sound/tas5720.txt +++ b/Documentation/devicetree/bindings/sound/tas5720.txt @@ -6,10 +6,12 @@ audio playback. For more product information please see the links below: http://www.ti.com/product/TAS5720L http://www.ti.com/product/TAS5720M +http://www.ti.com/product/TAS5722L Required properties: -- compatible : "ti,tas5720" +- compatible : "ti,tas5720", + "ti,tas5722" - reg : I2C slave address - dvdd-supply : phandle to a 3.3-V supply for the digital circuitry - pvdd-supply : phandle to a supply used for the Class-D amp and the analog diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index a736a2a6976c..5def54d1336d 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -36,6 +36,11 @@ /* Define how often to check (and clear) the fault status register (in ms) */ #define TAS5720_FAULT_CHECK_INTERVAL 200 +enum tas572x_type { + TAS5720, + TAS5722, +}; + static const char * const tas5720_supply_names[] = { "dvdd", /* Digital power supply. Connect to 3.3-V supply. */ "pvdd", /* Class-D amp and analog power supply (connected). */ @@ -47,6 +52,7 @@ struct tas5720_data { struct snd_soc_codec *codec; struct regmap *regmap; struct i2c_client *tas5720_client; + enum tas572x_type devtype; struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES]; struct delayed_work fault_check_work; unsigned int last_fault; @@ -264,7 +270,7 @@ static void tas5720_fault_check_work(struct work_struct *work) static int tas5720_codec_probe(struct snd_soc_codec *codec) { struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); - unsigned int device_id; + unsigned int device_id, expected_device_id; int ret; tas5720->codec = codec; @@ -276,6 +282,11 @@ static int tas5720_codec_probe(struct snd_soc_codec *codec) return ret; } + /* + * Take a liberal approach to checking the device ID to allow the + * driver to be used even if the device ID does not match, however + * issue a warning if there is a mismatch. + */ ret = regmap_read(tas5720->regmap, TAS5720_DEVICE_ID_REG, &device_id); if (ret < 0) { dev_err(codec->dev, "failed to read device ID register: %d\n", @@ -283,13 +294,22 @@ static int tas5720_codec_probe(struct snd_soc_codec *codec) goto probe_fail; } - if (device_id != TAS5720_DEVICE_ID) { - dev_err(codec->dev, "wrong device ID. expected: %u read: %u\n", - TAS5720_DEVICE_ID, device_id); - ret = -ENODEV; - goto probe_fail; + switch (tas5720->devtype) { + case TAS5720: + expected_device_id = TAS5720_DEVICE_ID; + break; + case TAS5722: + expected_device_id = TAS5722_DEVICE_ID; + break; + default: + dev_err(codec->dev, "unexpected private driver data\n"); + return -EINVAL; } + if (device_id != expected_device_id) + dev_warn(codec->dev, "wrong device ID. expected: %u read: %u\n", + expected_device_id, device_id); + /* Set device to mute */ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, TAS5720_MUTE, TAS5720_MUTE); @@ -552,6 +572,8 @@ static int tas5720_probe(struct i2c_client *client, return -ENOMEM; data->tas5720_client = client; + data->devtype = id->driver_data; + data->regmap = devm_regmap_init_i2c(client, &tas5720_regmap_config); if (IS_ERR(data->regmap)) { ret = PTR_ERR(data->regmap); @@ -592,7 +614,8 @@ static int tas5720_remove(struct i2c_client *client) } static const struct i2c_device_id tas5720_id[] = { - { "tas5720", 0 }, + { "tas5720", TAS5720 }, + { "tas5722", TAS5722 }, { } }; MODULE_DEVICE_TABLE(i2c, tas5720_id); @@ -600,6 +623,7 @@ MODULE_DEVICE_TABLE(i2c, tas5720_id); #if IS_ENABLED(CONFIG_OF) static const struct of_device_id tas5720_of_match[] = { { .compatible = "ti,tas5720", }, + { .compatible = "ti,tas5722", }, { }, }; MODULE_DEVICE_TABLE(of, tas5720_of_match); diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h index 3d077c779b12..bef802afcc69 100644 --- a/sound/soc/codecs/tas5720.h +++ b/sound/soc/codecs/tas5720.h @@ -32,6 +32,7 @@ /* TAS5720_DEVICE_ID_REG */ #define TAS5720_DEVICE_ID 0x01 +#define TAS5722_DEVICE_ID 0x12 /* TAS5720_POWER_CTRL_REG */ #define TAS5720_DIG_CLIP_MASK GENMASK(7, 2) -- 2.15.0

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[alsa-devel] ALSA device with slave clock
by Jussi Laako 15 Jan '18

15 Jan '18

Hi, I've been fighting with some devices for the case where ALSA device is a clock slave. I'm also working on some patches/quirks to make some of these things work. I have couple of questions in this domain I didn't find documented: 1) How is application supposed to use ALSA capture device when it doesn't know the sampling rate? For example with ASIO on Windows application sets sampling rate to 0 to indicate slave and then starts the device and retrieves the rate using ASIOGetSampleRate(). There are typically also reset request messages when the rate changes. 2) Some USB devices have indicator for lock status between set sampling rate and detected input rate. One such device I have is Motu 8D (UAC2 device): Simple mixer control 'Internal Clock Validity',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] As long as the status is valid, I can set sampling rate to the same, or something else. When the status becomes invalid due to rate mismatch, ALSA fails to set any of the sampling rates, even correct one and there's an error message in dmesg: "usb 3-14: clock source 1 is not valid, cannot use". So the clock source cannot be made valid again by changing sampling rate once it has become invalid, because it is currently invalid... 3) What is the correct way to deal with clock slave device from both driver and application perspective? So that application can retrieve the current sampling rate, instead of setting one... UAC2 has provisions for this, but this doesn't seem to be supported by the current driver. I have also some hardware where I would like to do the same with I2S - where clock generator is outside of the SoC and the SoC is just a slave (for example S/PDIF or AES/EBU input to a SoC where clock is generated from the input stream using PLL). Best regards, - Jussi

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