December 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: Intel: Change kern log level to avoid unwanted messages
by Naveen Manohar 08 Dec '17

08 Dec '17

patch suppresses the warning message "control load not supported" as this is a debug information to help debug issues in topology. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Acked-By: Vinod Koul <vinod.koul(a)intel.com> --- v1->v2: Added Intel sub-system sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a072bcf..81923da 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2908,7 +2908,7 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, break; default: - dev_warn(bus->dev, "Control load not supported %d:%d:%d\n", + dev_dbg(bus->dev, "Control load not supported %d:%d:%d\n", hdr->ops.get, hdr->ops.put, hdr->ops.info); break; } -- 1.9.1

1 0

[alsa-devel] [PATCH][next] ASoC: pcm186x: make pcm186x_dai_ops and pcm186x_adc_input_channel_sel_value static
by Colin King 07 Dec '17

07 Dec '17

From: Colin Ian King <colin.king(a)canonical.com> pcm186x_dai_ops and pcm186x_adc_input_channel_sel_value are local to the source and do not need to be in global scope, so make them static. Cleans up sparse warnings: symbol 'pcm186x_dai_ops' was not declared. Should it be static? symbol 'pcm186x_adc_input_channel_sel_value' was not declared. Should it be static? Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/pcm186x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index f7aa56e20169..cdb51427facc 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -59,7 +59,7 @@ static const struct snd_kcontrol_new pcm1865_snd_controls[] = { pcm186x_pga_tlv), }; -const unsigned int pcm186x_adc_input_channel_sel_value[] = { +static const unsigned int pcm186x_adc_input_channel_sel_value[] = { 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x20, 0x30 @@ -475,7 +475,7 @@ static int pcm186x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, return 0; } -const struct snd_soc_dai_ops pcm186x_dai_ops = { +static const struct snd_soc_dai_ops pcm186x_dai_ops = { .set_sysclk = pcm186x_set_dai_sysclk, .set_tdm_slot = pcm186x_set_tdm_slot, .set_fmt = pcm186x_set_fmt, -- 2.14.1

2 1

[alsa-devel] [PATCH v3 00/10] Add Headphone Detection to TLV320AIC31xx Driver
by Andrew F. Davis 07 Dec '17

07 Dec '17

Hello all, This series has the end goal of adding headphone detection to the tlv320aic31xx driver. The first few patches are mostly cleanups. Then a couple bug fixes I noticed. Followed by adding interrupt handling and finally headphone detection. Still not sure if the IRQ patch is correct, I think we have no good way to share this IRQ line, so leaving it the way it is for now. Thanks, Andrew Changes from v2: - Rebased on -next - Added delta patch for GPIO header fix - Split platform data remove into fwnode and pdata removal - Rework reset patch to print error code - Use snd_soc_codec_set_jack to set jack Changes from v1: - Splitup the cleanup patch a bit more - Move the GPIO1 register fix patch before header cleanup so it can be taken back into stable - Added Acked-by - New patch dealing with regulator notifications - Various small touchups - Rebased on v4.15-rc1 Andrew F. Davis (10): ASoC: tlv320aic31xx: Fix GPIO header includes ASoC: tlv320aic31xx: Use fwnode APIs over raw OF calls ASoC: tlv320aic31xx: Remove platform data ASoC: tlv320aic31xx: Add MICBIAS off setting ASoC: tlv320aic31xx: Remove regulator notification handling ASoC: tlv320aic31xx: Reset registers during power up ASoC: tlv320aic31xx: Add short circuit detection support ASoC: tlv320aic31xx: Add overflow detection support ASoC: tlv320aic31xx: Add headphone/headset detection ASoC: tlv320aic31xx: Add button press detection .../devicetree/bindings/sound/tlv320aic31xx.txt | 1 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 1 + sound/soc/codecs/tlv320aic31xx.c | 263 +++++++++++++-------- sound/soc/codecs/tlv320aic31xx.h | 41 +++- 4 files changed, 196 insertions(+), 110 deletions(-) -- 2.15.0

2 13

[alsa-devel] [PATCH v2 00/19] Add Headphone Detection to TLV320AIC31xx Driver
by Andrew F. Davis 07 Dec '17

07 Dec '17

Hello all, This series has the end goal of adding headphone detection to the tlv320aic31xx driver. The first few patches are mostly cleanups. Then a couple bug fixes I noticed. Followed by adding interrupt handling and finally headphone detection. This series (or at least patch #8) depend on this DT fix[0]. Thanks, Andrew Changes from v1: - Splitup the cleanup patch a bit more - Move the GPIO1 register fix patch before header cleanup so it can be taken back into stable - Added Acked-by - New patch dealing with regulator notifications - Various small touchups - Rebased on v4.15-rc1 + [0] [0] http://www.spinics.net/lists/kernel/msg2660968.html Andrew F. Davis (19): ASoC: tlv320aic31xx: File header and copyright cleanup ASoC: tlv320aic31xx: Change aic31xx_power_off return type to void ASoC: tlv320aic31xx: Move ACPI table next to OF table ASoC: tlv320aic31xx: General source formatting cleanup ASoC: tlv320aic31xx: Fix GPIO1 register definition ASoC: tlv320aic31xx: Reformat header file using GENMASK and BIT macros ASoC: tlv320aic31xx: Merge init function into probe ASoC: tlv320aic31xx: Switch GPIO handling to use gpiod_* API ASoC: tlv320aic31xx: Remove platform data ASoC: tlv320aic31xx: Add MICBIAS off setting ASoC: tlv320aic31xx: Check clock and divider before division ASoC: tlv320aic31xx: Add CODEC clock slave support ASoC: tlv320aic31xx: Fix inverted BCLK handling ASoC: tlv320aic31xx: Remove regulator notification handling ASoC: tlv320aic31xx: Reset registers during power up ASoC: tlv320aic31xx: Add short circuit detection support ASoC: tlv320aic31xx: Add overflow detection support ASoC: tlv320aic31xx: Add headphone/headset detection ASoC: tlv320aic31xx: Add button press detection .../devicetree/bindings/sound/tlv320aic31xx.txt | 1 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 1 + sound/soc/codecs/tlv320aic31xx.c | 422 ++++++++++-------- sound/soc/codecs/tlv320aic31xx.h | 496 ++++++++++----------- 4 files changed, 471 insertions(+), 449 deletions(-) rewrite sound/soc/codecs/tlv320aic31xx.h (83%) -- 2.15.0

2 62

[alsa-devel] [RFC 00/10] Enable HDA Codec support on Intel Platforms (Series2)
by Rakesh Ughreja 07 Dec '17

07 Dec '17

Many Intel platforms (SKL,KBL) etc. in the market supports enahanced audio capabilities which also includes DSP processing. This patch carry forwads the works that is done in the previous series to enable HD Audio codecs on such platforms. This patch series adds ASoC HDA codec driver for Intel platforms. It is written by reusing the legacy HDA ALSA codec driver. Intention is to maximize the reuse and minimize the changes in the legacy HDA codec driver. I would like to receive feedback before proceeding further on this direction. TODO: - This series is tested on KBL based product (Dell XPS 13). - Basic playback is working with headset and speakers. - Capture operation is not tested. - More platforms and use cases coverage can be added once we have basic agreement in terms of the overall approach. FIXME: - KConfig changes does not look right, but I could not think of any proper way without making changes into legacy HDA codec driver. So need some help on this topic. Rakesh Ughreja (10): ASoC: Intel: Boards: Machine driver for Intel platforms ASoC: Intel: Skylake: Add entry in sst_acpi_mach for HDA codecs ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ALSA: hda - make some of the functions externally visible ASoC: hdac_hda: add ASoC based HDA codec driver ALSA: hda: add new API snd_hda_asoc_codec_new for ASoC codec drivers ASoC: hdac_hda: add DAI, widgets and related ops ASoC: hdac_hda: add runtime PM support ASoC: Intel: Boards: add support for HDA codecs include/sound/hdaudio_ext.h | 3 +- sound/hda/ext/hdac_ext_bus.c | 9 +- sound/pci/hda/Kconfig | 11 + sound/pci/hda/hda_codec.c | 109 +++++- sound/pci/hda/hda_codec.h | 32 +- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 633 +++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 16 + sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_generic.c | 386 +++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 32 +- sound/soc/intel/skylake/skl.c | 79 +++- sound/soc/intel/skylake/skl.h | 10 +- 15 files changed, 1315 insertions(+), 26 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_generic.c -- 2.7.4

5 47

[alsa-devel] [PATCH] alsa-utils: aplay: Adjust sample rate limits to support newer hardware
by Jussi Laako 07 Dec '17

07 Dec '17

There are number of devices that support up to 384 kHz sampling rate and some devices up to 768 kHz sampling rate. This patch increases sanity check limit to 768k in order to support testing of such hardware. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- aplay/aplay.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index dbae17c..6b740c2 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -635,10 +635,10 @@ int main(int argc, char *argv[]) error(_("invalid rate argument '%s'"), optarg); return 1; } - if (tmp < 300) + if (tmp < 1000) tmp *= 1000; rhwparams.rate = tmp; - if (tmp < 2000 || tmp > 192000) { + if (tmp < 2000 || tmp > 768000) { error(_("bad speed value %i"), tmp); return 1; } -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add native DSD support for Esoteric D-05X
by Jussi Laako 07 Dec '17

07 Dec '17

Adds VID:PID of Esoteric D-05X to the TEAC device id's. Renames the is_teac_50X_dac() function to is_teac_dsd_dac() to cover broader device family from the same corporation sharing the same USB audio implementation. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 77eecaa..a66ef57 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1166,10 +1166,11 @@ static bool is_marantz_denon_dac(unsigned int id) /* TEAC UD-501/UD-503/NT-503 USB DACs need a vendor cmd to switch * between PCM/DOP and native DSD mode */ -static bool is_teac_50X_dac(unsigned int id) +static bool is_teac_dsd_dac(unsigned int id) { switch (id) { case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ + case USB_ID(0x0644, 0x8044): /* Esoteric D-05X */ return true; } return false; @@ -1202,7 +1203,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, break; } mdelay(20); - } else if (is_teac_50X_dac(subs->stream->chip->usb_id)) { + } else if (is_teac_dsd_dac(subs->stream->chip->usb_id)) { /* Vendor mode switch cmd is required. */ switch (fmt->altsetting) { case 3: /* DSD mode (DSD_U32) requested */ @@ -1392,7 +1393,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } /* TEAC devices with USB DAC functionality */ - if (is_teac_50X_dac(chip->usb_id)) { + if (is_teac_dsd_dac(chip->usb_id)) { if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } -- 2.7.4

2 1

[alsa-devel] [PATCH v2 0/6] ASoC: Intel: Skylake: Skylake driver updates
by Guneshwor Singh 07 Dec '17

07 Dec '17

This series includes fixes for HDMI widgets sysfs not updating, shim read optimization, fixes for dmic dev_type check and feature to configure DSP clock source. Guneshwor Singh (1): ASoC: Intel: Skylake: Do not check dev_type for dmic link type Pradeep Tewani (2): ASoC: Intel: Skylake: Parse vendor tokens to build A-State table ASoC: Intel: Skylake: Configure DSP clock source Puneeth Prabhu (2): ASoC: hdac_hdmi: Refresh sysfs during hdmi device probe ASoC: hdac_hdmi: Remove redundant assignments Subhransu S. Prusty (1): ASoC: Intel: Skylake: Remove second shim read in register_poll include/uapi/sound/snd_sst_tokens.h | 17 ++++++++++++- sound/soc/codecs/hdac_hdmi.c | 8 ++----- sound/soc/intel/common/sst-dsp.c | 4 +--- sound/soc/intel/skylake/skl-messages.c | 18 ++++++++++++++ sound/soc/intel/skylake/skl-nhlt.c | 15 ++++++++---- sound/soc/intel/skylake/skl-pcm.c | 6 +++++ sound/soc/intel/skylake/skl-sst-dsp.h | 3 +++ sound/soc/intel/skylake/skl-topology.c | 44 +++++++++++++++++++++++++++++++++- sound/soc/intel/skylake/skl.h | 17 +++++++++++++ 9 files changed, 116 insertions(+), 16 deletions(-) -- 2.15.1

4 12

[alsa-devel] Multiple codecs on the same DAI
by Ricard Wanderlof 07 Dec '17

07 Dec '17

Hi, I'm studying a hardware implementation where an S/PDIF receiver is on the same I2S connection as codec. Essentially, the S/PDIF receiver echos the I2S capture data from the codec unless there is a digital input signal which it can lock on to. For playback, the I2S output from the CPU DAI goes directly to the codec. I'm a bit at a loss for how to model this in ALSA. I know it's possible to have multiple codecs on a card, but those cases seem to be either separate codecs for the left and right channels, or separate codecs for capture and playback. In my case the playback codec is always the same, but the capture codec can switch between the two avilable input methods (more or less automatically if the S/PDIF receiver is so configured). >From a user perspective, it's not important that ALSA knows where the data is actually coming from, so if I could just register the two "codecs" on the same dai_link at the same time it would be fine, but I'm just imagining that I'll run into problems when ALSA tries to set up streams with two different codecs being involved. (Of course, at some level, the user might be interested to know the signal routing, but that can be determined by reading appropriate status bits in the S/PDIF receiver for instance using amixer controls). Do you have any ideas about how to configure this in ALSA? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 11

[alsa-devel] [PATCH v4] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 07 Dec '17

07 Dec '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later on each command. Otherwise, Playback volume will be 0dB. Basically, it should be 1. PowerDownRelease by Power Management1 <= call 1.x after 5LRCK 1.x Dummy write to Power Management1 2. PowerDownRelease by Power Management3 <= call 2.x after 5LRCK 2.x Dummy write to Power Management3 To avoid too many dummy write, this patch is merging these. 1. PowerDownRelease by Power Management1 2. PowerDownRelease by Power Management3 <= call after 5LRCK 2.x Dummy write to Power Management1/3 <= merge dummy write This patch adds dummy write when Playback Start timing. Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v3 -> v4 - use schedule_work() instead of schedule_delayed_work() sound/soc/codecs/ak4613.c | 60 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 60 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..8d2c90b 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct work_struct dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,65 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + /* wait 5 LR clocks */ + udelay(5000000 / priv->rate); + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delay/dummy_write method from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. + * + * But, unfortunately, we can't "write" here because here is atomic + * context (It uses I2C access for write). + * Thus, use delayed work with 0 delay to switch to normal context + * immediately, and wait 5 LR clocks and do dummy_write there. + */ + + if ((cmd != SNDRV_PCM_TRIGGER_START) && + (cmd != SNDRV_PCM_TRIGGER_RESUME)) + return 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + priv->component = &codec->component; + schedule_work(&priv->dummy_write_work); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +650,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1

2 4