March 2020 - Alsa-devel - mailman.alsa-project.org

[RFC PATCH] soundwire: bus: Add flag to mark DPN_BlockCtrl1 as readonly
by Srinivas Kandagatla 11 Mar '20

11 Mar '20

According to SoundWire Specification Version 1.2. "A Data Port number X (in the range 0-14) which supports only one value of WordLength may implement the WordLength field in the DPX_BlockCtrl1 Register as Read-Only, returning the fixed value of WordLength in response to reads." As WSA881x interfaces in PDM mode making the only field "WordLength" in DPX_BlockCtrl1" fixed and read-only. Behaviour of writing to this register on WSA881x soundwire slave with Qualcomm Soundwire Controller is throwing up an error. Not sure how other controllers deal with writing to readonly registers, but this patch provides a way to avoid writes to DPN_BlockCtrl1 register by providing a ro_blockctrl1_reg flag in struct sdw_port_runtime. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- I will send patch for WSA881x to include this change once this patch is accepted. drivers/soundwire/bus.h | 2 ++ drivers/soundwire/stream.c | 17 ++++++++++------- 2 files changed, 12 insertions(+), 7 deletions(-) diff --git a/drivers/soundwire/bus.h b/drivers/soundwire/bus.h index 204204a26db8..791e8d14093e 100644 --- a/drivers/soundwire/bus.h +++ b/drivers/soundwire/bus.h @@ -79,6 +79,7 @@ int sdw_find_col_index(int col); * @num: Port number. For audio streams, valid port number ranges from * [1,14] * @ch_mask: Channel mask + * @ro_blockctrl1_reg: Read Only flag for DPN_BlockCtrl1 register * @transport_params: Transport parameters * @port_params: Port parameters * @port_node: List node for Master or Slave port_list @@ -89,6 +90,7 @@ int sdw_find_col_index(int col); struct sdw_port_runtime { int num; int ch_mask; + bool ro_blockctrl1_reg; struct sdw_transport_params transport_params; struct sdw_port_params port_params; struct list_head port_node; diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 00348d1fc606..4491643aeb4a 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -167,13 +167,15 @@ static int sdw_program_slave_port_params(struct sdw_bus *bus, return ret; } - /* Program DPN_BlockCtrl1 register */ - ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1)); - if (ret < 0) { - dev_err(&s_rt->slave->dev, - "DPN_BlockCtrl1 register write failed for port %d\n", - t_params->port_num); - return ret; + if (!p_rt->ro_blockctrl1_reg) { + /* Program DPN_BlockCtrl1 register */ + ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1)); + if (ret < 0) { + dev_err(&s_rt->slave->dev, + "DPN_BlockCtrl1 register write failed for port %d\n", + t_params->port_num); + return ret; + } } /* Program DPN_SampleCtrl1 register */ @@ -1195,6 +1197,7 @@ static struct sdw_port_runtime p_rt->ch_mask = port_config[port_index].ch_mask; p_rt->num = port_config[port_index].num; + p_rt->ro_blockctrl1_reg = port_config[port_index].ro_blockctrl1_reg; return p_rt; } -- 2.21.0

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[PATCH] ALSA: hda/realtek: Fix pop noise on ALC225
by Kai-Heng Feng 11 Mar '20

11 Mar '20

Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. So partially revert this commit for ALC225 to fix the regression. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ac06ff1a17c..7b83b020ac3c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8051,6 +8051,8 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: + codec->power_save_node = 1; + /* fall through */ case 0x10ec0295: case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; -- 2.17.1

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[PATCH] alsaloop: reduce cumulative error caused by non-atomic samples calculation
by Ruslan Bilovol 11 Mar '20

11 Mar '20

When doing loopback between two audio card with same sampling frequency, I noticed slow increase of pitch_diff. When I changed order of get_queued_playback_samples() vs get_queued_capture_samples(), I noticed same drift of pitch_diff but if was decreasing this time. This seems to be caused by non-atomic consecutive snd_pcm_delay() invocation for playback then for capture. snd_pcm_delay() measures delay between read/write call and actual ADC/DAC operation. So while we get this value for playback path in get_queued_playback_samples(), next call to get_queued_capture_samples() will happen a little bit later so snd_pcm_delay() may return incorrect value. Be interleaving get_queued_{playback,capture}_samples() order, we divide this small error between playback and capture paths. I do not see any issues anymore with one-way drift of pitch_diff. Signed-off-by: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> --- alsaloop/pcmjob.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index b252486..1b7925a 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1951,8 +1951,16 @@ int pcmjob_pollfds_handle(struct loopback *loop, struct pollfd *fds) } if (loop->sync != SYNC_TYPE_NONE) { snd_pcm_sframes_t pqueued, cqueued; - pqueued = get_queued_playback_samples(loop); - cqueued = get_queued_capture_samples(loop); + + /* Reduce cumulative error by interleaving playback vs capture reading order */ + if (loop->total_queued_count & 1) { + pqueued = get_queued_playback_samples(loop); + cqueued = get_queued_capture_samples(loop); + } else { + cqueued = get_queued_capture_samples(loop); + pqueued = get_queued_playback_samples(loop); + } + if (verbose > 4) snd_output_printf(loop->output, "%s: queued %li/%li samples\n", loop->id, pqueued, cqueued); if (pqueued > 0) -- 1.9.1

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[PATCH 7/8] ALSA: firewire-tascam: Add missing annotation for tscm_hwdep_read_queue()
by Jules Irenge 11 Mar '20

11 Mar '20

Sparse reports a warning at tscm_hwdep_read_queue() warning: context imbalance in tscm_hwdep_read_queue() - unexpected unlock The root cause is the missing annotation at tscm_hwdep_read_queue() Add the missing __releases(&tscm->lock) annotation Signed-off-by: Jules Irenge <jbi.octave(a)gmail.com> --- sound/firewire/tascam/tascam-hwdep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index c29a97f6f638..9801e33e7f2a 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -36,6 +36,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf, long remained, loff_t *offset) + __releases(&tscm->lock) { char __user *pos = buf; unsigned int type = SNDRV_FIREWIRE_EVENT_TASCAM_CONTROL; -- 2.24.1

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[PATCH 8/8] ALSA: firewire-tascam: Add missing annotation for tscm_hwdep_read_locked()
by Jules Irenge 11 Mar '20

11 Mar '20

Sparse reports a warning at tscm_hwdep_read_locked() warning: context imbalance in tscm_hwdep_read_locked() - unexpected unlock The root cause is the missing annotation at tscm_hwdep_read_locked() Add the missing __releases(&tscm->lock) annotation Signed-off-by: Jules Irenge <jbi.octave(a)gmail.com> --- sound/firewire/tascam/tascam-hwdep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index 9801e33e7f2a..6f38335fe10b 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -17,6 +17,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, long count, loff_t *offset) + __releases(&tscm->lock) { struct snd_firewire_event_lock_status event = { .type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS, -- 2.24.1

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Sample project request...
by GitHub issues - opened 11 Mar '20

11 Mar '20

alsa-project/alsa-lib issue #35 was opened from mnini: Can you make sample of alsa processor that is have input and output in the code? I know only syntax of C and I want to program DSP for linux output sound. Issue URL : https://github.com/alsa-project/alsa-lib/issues/35 Repository URL: https://github.com/alsa-project/alsa-lib

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How to model 3xADAU1761 codec in TDM8 with 2xSAI in ALSA ?
by Rolf Peder Klemetsen 10 Mar '20

10 Mar '20

Hello, I am working on bringup of a custom board that has three ADAU1761 codecs, and a NXP LS1028A SoC. The codecs are supposed to be configured in TDM8 mode and share a common playback I2S bus connected to SAI4, and a common capture I2S bus connected to SAI3. What is the recommended way of modelling this setup in the Linux device tree ? So far I have tried the following: &i2c0 { status = "okay"; i2c_codec1: adau1761@38 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x38>; tdm-offset = <0>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; i2c_codec2: adau1761@39 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x39>; tdm-offset = <16>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; i2c_codec3: adau1761@3a { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x3a>; tdm-offset = <32>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; }; At first I attempted creating three simple-audio-card's. That attempted failed since sharing the same cpu (SAI3 and SAI4) failed with no ALSA device as a result. Then I attempted creating a single simple-audio-card with two dai-links (one for playback and one for capture), each containing three codec sub-nodes. That doesn't really work either. Can someone please point me in the right direction? Many of the parameters that I have put in the dts have no effect and are assumed not supported. Regards Rolf sound { status = "okay"; model = "Super Sound System"; compatible = "simple-audio-card"; simple-audio-card,name = "sound"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&p_snd_codec_o1>; simple-audio-card,frame-master = <&p_snd_codec_o1>; simple-audio-card,widgets = "Microphone", "Mic In", "Headphone", "Headphone Out", "Line", "Line In", "Line", "Line Out"; simple-audio-card,routing = "Line Out", "LOUT", "Line Out", "ROUT", "Headphone Out", "LHP", "Headphone Out", "RHP", "Mic In", "MICBIAS", "LINN", "Mic In", "RINN", "Mic In", "LINP", "Mic In", "RINP", "Mic In", "LAUX", "Line In", "RAUX", "Line In"; playback_link: simple-audio-card,dai-link@0 { format = "i2s"; convert-channels = <8>; bitclock-master = <&p_snd_codec_o1>; frame-master = <&p_snd_codec_o1>; name-prefix = "p"; snd_dai_playback: cpu@0 { cpu-dai-name = "sai4_0"; sound-dai-name = "sai4_0"; sound-dai = <&sai4 0>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <1 0 1 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c1"; dai-tdm-slot-tx-mask = <1 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 1 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; }; capture_link: simple-audio-card,dai-link@1 { format = "i2s"; convert-channels = <8>; bitclock-master = <&c_snd_codec_o1>; frame-master = <&c_snd_codec_o1>; name-prefix = "c"; snd_dai_capture: cpu@0 { sound-dai = <&sai3>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 1 0 1 0 0>; }; c_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c1"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 0 0 0 0 0>; }; c_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 1 0 0 0 0>; }; c_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 1 0 0>; }; };

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[PATCH 0/3] ASoC: rt5682: fix Kconfig/compilation issues when I2C is not selected
by Pierre-Louis Bossart 10 Mar '20

10 Mar '20

RT5682 can work in I2C or SoundWire mode, but the code will not compile if I2C is not selected. Reported-by: kbuild test robot <lkp(a)intel.com> Pierre-Louis Bossart (3): ASoC: rt5682: fix unmet dependencies ASoC: rt5682: fix compilation issues without I2C ASoC: rt5682-sdw: fix 'defined but not used' pm functions sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/rt5682-sdw.c | 4 ++-- sound/soc/codecs/rt5682.c | 10 ++++++++++ 3 files changed, 13 insertions(+), 3 deletions(-) -- 2.20.1

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Re: [PATCH] dt-bindings: sound: Convert cirrus, cs42l51 to json-schema
by Rob Herring 10 Mar '20

10 Mar '20

On Fri, 28 Feb 2020 16:27:06 +0100, Benjamin Gaignard wrote: > Convert cirrus,cs42l51 to yaml format. > > Signed-off-by: Benjamin Gaignard <benjamin.gaignard(a)st.com> > --- > .../devicetree/bindings/sound/cirrus,cs42l51.yaml | 69 ++++++++++++++++++++++ > .../devicetree/bindings/sound/cs42l51.txt | 33 ----------- > 2 files changed, 69 insertions(+), 33 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml > delete mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt > Reviewed-by: Rob Herring <robh(a)kernel.org>

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[PATCH for-5.7] ASoC: pcm: Fix (again) possible buffer overflow in dpcm state sysfs output
by Takashi Iwai 10 Mar '20

10 Mar '20

This is re-applying the fix that went into 5.6 (commit 6c89ffea60aa) as the changes were wiped out after merging the other code refactoring. Basically the same changes, just replacing the suspicious calls of snprintf() with scnprintf(). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-pcm.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e7915adaaf0a..34ee54ea7b30 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -66,16 +66,16 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, unsigned long flags; /* FE state */ - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "[%s - %s]\n", fe->dai_link->name, stream ? "Capture" : "Playback"); - offset += snprintf(buf + offset, size - offset, "State: %s\n", + offset += scnprintf(buf + offset, size - offset, "State: %s\n", dpcm_state_string(fe->dpcm[stream].state)); if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "Hardware Params: " "Format = %s, Channels = %d, Rate = %d\n", snd_pcm_format_name(params_format(params)), @@ -83,10 +83,10 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_rate(params)); /* BEs state */ - offset += snprintf(buf + offset, size - offset, "Backends:\n"); + offset += scnprintf(buf + offset, size - offset, "Backends:\n"); if (list_empty(&fe->dpcm[stream].be_clients)) { - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " No active DSP links\n"); goto out; } @@ -96,16 +96,16 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "- %s\n", be->dai_link->name); - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " State: %s\n", dpcm_state_string(be->dpcm[stream].state)); if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " Hardware Params: " "Format = %s, Channels = %d, Rate = %d\n", snd_pcm_format_name(params_format(params)), -- 2.16.4

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