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

[alsa-devel] Applied "ASoC: adau1761: Use the standard fall-through annotation" to the asoc tree
by Mark Brown 05 Oct '18

05 Oct '18

The patch ASoC: adau1761: Use the standard fall-through annotation 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 7e29317928bd79a03a9c35816afa709988b5d31b Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Thu, 4 Oct 2018 20:30:02 +0200 Subject: [PATCH] ASoC: adau1761: Use the standard fall-through annotation As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, replace with the standard "fall through" annotation at the right place. It has to be put right before the next label. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau1761.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index be136e981653..bef3e9e74c26 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component) ARRAY_SIZE(adau1761_jack_detect_controls)); if (ret) return ret; - case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */ + /* fall through */ + case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes, ARRAY_SIZE(adau1761_no_dmic_routes)); if (ret) -- 2.19.0.rc2

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[alsa-devel] [PATCH 0/5] ASoC: Fix fall-through annotations
by Takashi Iwai 05 Oct '18

05 Oct '18

Hi, this is a patch set to add / fix the "fall through" annotations to be compliant with gcc's -Wimplicit-fallthrough option. Takashi === Takashi Iwai (5): ASoC: adau1761: Use the standard fall-through annotation ASoC: pcm186x: Use the standard fall-through annotation ASoC: rt274: Add fall-through annotations ASoC: intel: skylake: Add fall-through annotation ASoC: topology: Use the standard fall-through annotations sound/soc/codecs/adau1761.c | 3 ++- sound/soc/codecs/pcm186x.c | 3 ++- sound/soc/codecs/rt274.c | 2 ++ sound/soc/intel/skylake/skl-pcm.c | 1 + sound/soc/soc-topology.c | 4 ++-- 5 files changed, 9 insertions(+), 4 deletions(-) -- 2.18.0

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[alsa-devel] Applied "ASoC: intel: skylake: Add fall-through annotation" to the asoc tree
by Mark Brown 05 Oct '18

05 Oct '18

The patch ASoC: intel: skylake: Add fall-through annotation 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 e4bfd61571f5db5c69a7a49de401543cc7d6c87c Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Thu, 4 Oct 2018 20:30:05 +0200 Subject: [PATCH] ASoC: intel: skylake: Add fall-through annotation As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, add the "fall through" annotation in Intel SST skylake driver. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 00b7a91b18c9..557f80c0bfe5 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -495,6 +495,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -- 2.19.0.rc2

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[alsa-devel] [PATCH] ASoC: dt-bindings: add dt bindings for adau1977 audio codec
by Alexandru Ardelean 05 Oct '18

05 Oct '18

Add device-tree bindings documentation file for ADAU1977 audio codec. This describes device-tree fields that are already supported by the driver. The driver supports both I2C AND and SPI, and this doc covers both aspects of the DT configuration. Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- .../bindings/sound/adi,adau1977.txt | 54 +++++++++++++++++++ 1 file changed, 54 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1977.txt diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt new file mode 100644 index 000000000000..e79aeef73f28 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt @@ -0,0 +1,54 @@ +Analog Devices ADAU1977/ADAU1978/ADAU1979 + +Datasheets: +http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf +http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf +http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf + +This driver supports both the I2C and SPI bus. + +Required properties: + - compatible: Should contain one of the following: + "adi,adau1977" + "adi,adau1978" + "adi,adau1979" + + - AVDD-supply: analog power supply for the device, please consult + Documentation/devicetree/bindings/regulator/regulator.txt + +Optional properties: + - reset-gpio: the reset pin for the chip, for more details consult + Documentation/devicetree/bindings/gpio/gpio.txt + + - DVDD-supply: supply voltage for the digital core, please consult + Documentation/devicetree/bindings/regulator/regulator.txt + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: The i2c address. Value depends on the state of ADDR0 + and ADDR1, as wired in hardware. + +Examples: + + adau1977_spi: adau1977@0 { + compatible = "adi,adau1977"; + spi-max-frequency = <600000>; + + AVDD-supply = <&regulator>; + DVDD-supply = <&regulator_digital>; + + reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>; + }; + + adau1977_i2c: adau1977@11 { + compatible = "adi,adau1977"; + reg = <0x11>; + + AVDD-supply = <&regulator>; + DVDD-supply = <&regulator_digital>; + + reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>; + }; -- 2.17.1

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[alsa-devel] [PATCH v5] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

Adds several vendor specific mixer quirks for RME's Class Compliant USB devices. These provide extra status information from the device otherwise not available. These include AES/SPDIF rate and status information, current system sampling rate and measured frequency. This information is especially useful in cases where device's clock is slaved to external clock source. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/mixer_quirks.c | 381 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 381 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index cbfb48bdea51..85ae0ff2382a 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -29,6 +29,7 @@ #include <linux/hid.h> #include <linux/init.h> +#include <linux/math64.h> #include <linux/slab.h> #include <linux/usb.h> #include <linux/usb/audio.h> @@ -1817,6 +1818,380 @@ static int dell_dock_mixer_init(struct usb_mixer_interface *mixer) return 0; } +/* RME Class Compliant device quirks */ + +#define SND_RME_GET_STATUS1 23 +#define SND_RME_GET_CURRENT_FREQ 17 +#define SND_RME_CLK_SYSTEM_SHIFT 16 +#define SND_RME_CLK_SYSTEM_MASK 0x1f +#define SND_RME_CLK_AES_SHIFT 8 +#define SND_RME_CLK_SPDIF_SHIFT 12 +#define SND_RME_CLK_AES_SPDIF_MASK 0xf +#define SND_RME_CLK_SYNC_SHIFT 6 +#define SND_RME_CLK_SYNC_MASK 0x3 +#define SND_RME_CLK_FREQMUL_SHIFT 18 +#define SND_RME_CLK_FREQMUL_MASK 0x7 +#define SND_RME_CLK_SYSTEM(x) \ + ((x >> SND_RME_CLK_SYSTEM_SHIFT) & SND_RME_CLK_SYSTEM_MASK) +#define SND_RME_CLK_AES(x) \ + ((x >> SND_RME_CLK_AES_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SPDIF(x) \ + ((x >> SND_RME_CLK_SPDIF_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SYNC(x) \ + ((x >> SND_RME_CLK_SYNC_SHIFT) & SND_RME_CLK_SYNC_MASK) +#define SND_RME_CLK_FREQMUL(x) \ + ((x >> SND_RME_CLK_FREQMUL_SHIFT) & SND_RME_CLK_FREQMUL_MASK) +#define SND_RME_CLK_AES_LOCK 0x1 +#define SND_RME_CLK_AES_SYNC 0x4 +#define SND_RME_CLK_SPDIF_LOCK 0x2 +#define SND_RME_CLK_SPDIF_SYNC 0x8 +#define SND_RME_SPDIF_IF_SHIFT 4 +#define SND_RME_SPDIF_FORMAT_SHIFT 5 +#define SND_RME_BINARY_MASK 0x1 +#define SND_RME_SPDIF_IF(x) \ + ((x >> SND_RME_SPDIF_IF_SHIFT) & SND_RME_BINARY_MASK) +#define SND_RME_SPDIF_FORMAT(x) \ + ((x >> SND_RME_SPDIF_FORMAT_SHIFT) & SND_RME_BINARY_MASK) + +static const u32 snd_rme_rate_table[] = { + 32000, 44100, 48000, 50000, + 64000, 88200, 96000, 100000, + 128000, 176400, 192000, 200000, + 256000, 352800, 384000, 400000, + 512000, 705600, 768000, 800000 +}; +/* maximum number of items for AES and S/PDIF rates for above table */ +#define SND_RME_RATE_IDX_AES_SPDIF_NUM 12 + +enum snd_rme_domain { + SND_RME_DOMAIN_SYSTEM, + SND_RME_DOMAIN_AES, + SND_RME_DOMAIN_SPDIF +}; + +enum snd_rme_clock_status { + SND_RME_CLOCK_NOLOCK, + SND_RME_CLOCK_LOCK, + SND_RME_CLOCK_SYNC +}; + +static int snd_rme_read_value(struct snd_usb_audio *chip, + unsigned int item, + u32 *value) +{ + struct usb_device *dev = chip->dev; + int err; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + item, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0, 0, + value, sizeof(*value)); + if (err < 0) + dev_err(&dev->dev, + "unable to issue vendor read request %d (ret = %d)", + item, err); + return err; +} + +static int snd_rme_get_status1(struct snd_kcontrol *kcontrol, + u32 *status1) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, status1); + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + u32 rate = 0; + int idx; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + idx = SND_RME_CLK_SYSTEM(status1); + if (idx < ARRAY_SIZE(snd_rme_rate_table)) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_AES: + idx = SND_RME_CLK_AES(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_SPDIF: + idx = SND_RME_CLK_SPDIF(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + default: + return -EINVAL; + } + ucontrol->value.integer.value[0] = rate; + return 0; +} + +static int snd_rme_sync_state_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int idx = SND_RME_CLOCK_NOLOCK; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_AES: /* AES */ + if (status1 & SND_RME_CLK_AES_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_AES_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + case SND_RME_DOMAIN_SPDIF: /* SPDIF */ + if (status1 & SND_RME_CLK_SPDIF_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_SPDIF_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + default: + return -EINVAL; + } + ucontrol->value.enumerated.item[0] = idx; + return 0; +} + +static int snd_rme_spdif_if_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_IF(status1); + return 0; +} + +static int snd_rme_spdif_format_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_FORMAT(status1); + return 0; +} + +static int snd_rme_sync_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_CLK_SYNC(status1); + return 0; +} + +static int snd_rme_current_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + u32 status1; + const u64 num = 104857600000000ULL; + u32 den; + unsigned int freq; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, &status1); + if (err < 0) + goto end; + err = snd_rme_read_value(chip, SND_RME_GET_CURRENT_FREQ, &den); + if (err < 0) + goto end; + freq = (den == 0) ? 0 : div64_u64(num, den); + freq <<= SND_RME_CLK_FREQMUL(status1); + ucontrol->value.integer.value[0] = freq; + +end: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + uinfo->value.integer.min = 32000; + uinfo->value.integer.max = 800000; + break; + case SND_RME_DOMAIN_AES: + case SND_RME_DOMAIN_SPDIF: + default: + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 200000; + } + uinfo->value.integer.step = 0; + return 0; +} + +static int snd_rme_sync_state_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_states[] = { + "No Lock", "Lock", "Sync" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_states), sync_states); +} + +static int snd_rme_spdif_if_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const spdif_if[] = { + "Coaxial", "Optical" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(spdif_if), spdif_if); +} + +static int snd_rme_spdif_format_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const optical_type[] = { + "Consumer", "Professional" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(optical_type), optical_type); +} + +static int snd_rme_sync_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_sources[] = { + "Internal", "AES", "SPDIF", "Internal" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_sources), sync_sources); +} + +static struct snd_kcontrol_new snd_rme_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Interface", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_if_info, + .get = snd_rme_spdif_if_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_format_info, + .get = snd_rme_spdif_format_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sync Source", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_source_info, + .get = snd_rme_sync_source_get + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "System Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SYSTEM + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Current Frequency", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_current_freq_get + } +}; + +static int snd_rme_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_rme_controls); ++i) { + err = add_single_ctl_with_resume(mixer, 0, + NULL, + &snd_rme_controls[i], + NULL); + if (err < 0) + return err; + } + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1904,6 +2279,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ err = dell_dock_mixer_init(mixer); break; + + case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */ + case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */ + case USB_ID(0x2a39, 0x3fd4): /* RME */ + err = snd_rme_controls_create(mixer); + break; } return err; -- 2.17.1

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[alsa-devel] [PATCH v2 0/5] ASoC: qdsp6: add compress offload support
by Srinivas Kandagatla 05 Oct '18

05 Oct '18

This patchset adds support to very basic MP3 playback compress offload support via QDSP ASM module. Tested this patchset on DB410c with APQ8016 and DB820c with APQ8096 using tinycompress library. Adding other codec support should be trivial w.r.t qdsp6 side, however there are other dependencies like compress UAPI header changes and FastRPC which are being worked in parallel. Once ready will post them! thanks, srini Changes since v1 (https://patchwork.kernel.org/cover/10585805/) - added support to snd_compr_set_runtime_buffer() - make use of generic copy function - few trivial cleanups suggested by Vinod Koul Srinivas Kandagatla (5): ALSA: soc-compress: add support to snd_compr_set_runtime_buffer() ALSA: compress: make use of runtime buffer for copy ASoC: q6asm-dai: dt-bindings: Add support to compress dais ASoC: qdsp6: q6asm: add support to MP3 format ASoC: qdsp6: q6asm-dai: Add support to compress offload .../devicetree/bindings/sound/qcom,q6asm.txt | 27 ++ include/sound/compress_driver.h | 19 + sound/core/compress_offload.c | 18 +- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 377 +++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 5 + 6 files changed, 443 insertions(+), 4 deletions(-) -- 2.19.0

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[alsa-devel] [PATCH v4] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

Adds several vendor specific mixer quirks for RME's Class Compliant USB devices. These provide extra status information from the device otherwise not available. These include AES/SPDIF rate and status information, current system sampling rate and measured frequency. This information is especially useful in cases where device's clock is slaved to external clock source. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/mixer_quirks.c | 380 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 380 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index cbfb48bdea51..6988dc21c31e 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -29,6 +29,7 @@ #include <linux/hid.h> #include <linux/init.h> +#include <linux/math64.h> #include <linux/slab.h> #include <linux/usb.h> #include <linux/usb/audio.h> @@ -1817,6 +1818,379 @@ static int dell_dock_mixer_init(struct usb_mixer_interface *mixer) return 0; } +/* RME Class Compliant device quirks */ + +#define SND_RME_GET_STATUS1 23 +#define SND_RME_GET_CURRENT_FREQ 17 +#define SND_RME_CLK_SYSTEM_SHIFT 16 +#define SND_RME_CLK_SYSTEM_MASK 0x1f +#define SND_RME_CLK_AES_SHIFT 8 +#define SND_RME_CLK_SPDIF_SHIFT 12 +#define SND_RME_CLK_AES_SPDIF_MASK 0xf +#define SND_RME_CLK_SYNC_SHIFT 6 +#define SND_RME_CLK_SYNC_MASK 0x3 +#define SND_RME_CLK_FREQMUL_SHIFT 18 +#define SND_RME_CLK_FREQMUL_MASK 0x7 +#define SND_RME_CLK_SYSTEM(x) \ + ((x >> SND_RME_CLK_SYSTEM_SHIFT) & SND_RME_CLK_SYSTEM_MASK) +#define SND_RME_CLK_AES(x) \ + ((x >> SND_RME_CLK_AES_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SPDIF(x) \ + ((x >> SND_RME_CLK_SPDIF_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SYNC(x) \ + ((x >> SND_RME_CLK_SYNC_SHIFT) & SND_RME_CLK_SYNC_MASK) +#define SND_RME_CLK_FREQMUL(x) \ + ((x >> SND_RME_CLK_FREQMUL_SHIFT) & SND_RME_CLK_FREQMUL_MASK) +#define SND_RME_CLK_AES_LOCK 0x1 +#define SND_RME_CLK_AES_SYNC 0x4 +#define SND_RME_CLK_SPDIF_LOCK 0x2 +#define SND_RME_CLK_SPDIF_SYNC 0x8 +#define SND_RME_SPDIF_IF_SHIFT 4 +#define SND_RME_SPDIF_FORMAT_SHIFT 5 +#define SND_RME_BINARY_MASK 0x1 +#define SND_RME_SPDIF_IF(x) \ + ((x >> SND_RME_SPDIF_IF_SHIFT) & SND_RME_BINARY_MASK) +#define SND_RME_SPDIF_FORMAT(x) \ + ((x >> SND_RME_SPDIF_FORMAT_SHIFT) & SND_RME_BINARY_MASK) + +static const u32 snd_rme_rate_table[] = { + 32000, 44100, 48000, 50000, + 64000, 88200, 96000, 100000, + 128000, 176400, 192000, 200000, + 256000, 352800, 384000, 400000, + 512000, 705600, 768000, 800000 +}; +#define SND_RME_RATE_IDX_AES_SPDIF_NUM 12 + +enum snd_rme_domain { + SND_RME_DOMAIN_SYSTEM, + SND_RME_DOMAIN_AES, + SND_RME_DOMAIN_SPDIF +}; + +enum snd_rme_clock_status { + SND_RME_CLOCK_NOLOCK, + SND_RME_CLOCK_LOCK, + SND_RME_CLOCK_SYNC +}; + +static int snd_rme_read_value(struct snd_usb_audio *chip, + unsigned int item, + u32 *value) +{ + struct usb_device *dev = chip->dev; + int err; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + item, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0, 0, + value, sizeof(*value)); + if (err < 0) + dev_err(&dev->dev, + "unable to issue vendor read request %d (ret = %d)", + item, err); + return err; +} + +static int snd_rme_get_status1(struct snd_kcontrol *kcontrol, + u32 *status1) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, status1); + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + u32 rate = 0; + int idx; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + idx = SND_RME_CLK_SYSTEM(status1); + if (idx < ARRAY_SIZE(snd_rme_rate_table)) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_AES: + idx = SND_RME_CLK_AES(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_SPDIF: + idx = SND_RME_CLK_SPDIF(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + default: + return -EINVAL; + } + ucontrol->value.integer.value[0] = rate; + return 0; +} + +static int snd_rme_sync_state_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int idx = SND_RME_CLOCK_NOLOCK; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_AES: /* AES */ + if (status1 & SND_RME_CLK_AES_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_AES_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + case SND_RME_DOMAIN_SPDIF: /* SPDIF */ + if (status1 & SND_RME_CLK_SPDIF_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_SPDIF_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + default: + return -EINVAL; + } + ucontrol->value.enumerated.item[0] = idx; + return 0; +} + +static int snd_rme_spdif_if_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_IF(status1); + return 0; +} + +static int snd_rme_spdif_format_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_FORMAT(status1); + return 0; +} + +static int snd_rme_sync_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_CLK_SYNC(status1); + return 0; +} + +static int snd_rme_current_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + u32 status1; + const u64 num = 104857600000000ULL; + u32 den; + unsigned int freq; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, &status1); + if (err < 0) + goto end; + err = snd_rme_read_value(chip, SND_RME_GET_CURRENT_FREQ, &den); + if (err < 0) + goto end; + freq = (den == 0) ? 0 : div64_u64(num, den); + freq <<= SND_RME_CLK_FREQMUL(status1); + ucontrol->value.integer.value[0] = freq; + +end: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + uinfo->value.integer.min = 32000; + uinfo->value.integer.max = 800000; + break; + case SND_RME_DOMAIN_AES: + case SND_RME_DOMAIN_SPDIF: + default: + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 200000; + } + uinfo->value.integer.step = 0; + return 0; +} + +static int snd_rme_sync_state_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_states[] = { + "No Lock", "Lock", "Sync" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_states), sync_states); +} + +static int snd_rme_spdif_if_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const spdif_if[] = { + "Coaxial", "Optical" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(spdif_if), spdif_if); +} + +static int snd_rme_spdif_format_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const optical_type[] = { + "Consumer", "Professional" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(optical_type), optical_type); +} + +static int snd_rme_sync_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_sources[] = { + "Internal", "AES", "SPDIF", "Internal" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_sources), sync_sources); +} + +static struct snd_kcontrol_new snd_rme_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Interface", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_if_info, + .get = snd_rme_spdif_if_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_format_info, + .get = snd_rme_spdif_format_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sync Source", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_source_info, + .get = snd_rme_sync_source_get + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "System Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SYSTEM + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Current Frequency", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_current_freq_get + } +}; + +static int snd_rme_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_rme_controls); ++i) { + err = add_single_ctl_with_resume(mixer, 0, + NULL, + &snd_rme_controls[i], + NULL); + if (err < 0) + return err; + } + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1904,6 +2278,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ err = dell_dock_mixer_init(mixer); break; + + case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */ + case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */ + case USB_ID(0x2a39, 0x3fd4): /* RME */ + err = snd_rme_controls_create(mixer); + break; } return err; -- 2.17.1

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[alsa-devel] [PATCH v3 0/1] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

Here are the requested updates. I have kept the locking separate from the helper, because in frequency calculation result depends on both values. I can move locking to the helper too if necessary, but I thought it would be logically cleaner this way...

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[alsa-devel] [PATCH] ALSA: usb-audio: update quirk for B&W PX to remove microphone
by Nicolas Huaman 05 Oct '18

05 Oct '18

A quirk in snd-usb-audio was added to automate setting sample rate to 4800k and remove the previously exposed nonfunctional microphone for the Bowers & Wilkins PX: commit 240a8af929c7c57dcde28682725b29cf8474e8e5 https://lore.kernel.org/patchwork/patch/919689/ However the headphones where updated shortly after that to remove the unintentional microphone functionality. I guess because of this the headphones now crash when connecting them via USB while the quirk is active. Dmesg: snd-usb-audio: probe of 2-3:1.0 failed with error -22 usb 2-3: 2:1: cannot get min/max values for control 2 (id 2) This patch removes the microfone and allows the headphones to connect and work out of the box. It is based on the current mainline kernel and successfully applied an tested on my machine (4.18.10.arch1-1). Signed-off-by: Nicolas Huaman <nicolas(a)herochao.de> --- sound/usb/quirks-table.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 08aa78007020..849953e5775c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3346,19 +3346,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | -- 2.19.0

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[alsa-devel] [PATCH v2 0/1] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

This patch has been updated with following changes and should be now good for merging: - Replace 64-bit division with suitable function to make it available on all architectures - Use specific VID:PID pairs instead of just VID to avoid confusions Last item was changed after discussion with RME.

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