April 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v4] sound: rawmidi: Add framing mode
by David Henningsson 13 Apr '21

13 Apr '21

This commit adds a new framing mode that frames all MIDI data into 32-byte frames with a timestamp. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Testing on a Celeron N3150 with this mode has a max jitter of 2.8 ms, compared to the in-kernel seq implementation which has a max jitter of 5 ms during idle and much worse when running scheduler stress tests in parallel. Signed-off-by: David Henningsson <coding(a)diwic.se> --- include/sound/rawmidi.h | 2 ++ include/uapi/sound/asound.h | 26 ++++++++++++++-- sound/core/rawmidi.c | 60 +++++++++++++++++++++++++++++++++++-- 3 files changed, 84 insertions(+), 4 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..b0057a193c31 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,8 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + u8 framing; /* whether to frame input data */ + clockid_t clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..af8e60740218 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -736,12 +736,34 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +enum { + SNDRV_RAWMIDI_FRAMING_NONE = 0, + SNDRV_RAWMIDI_FRAMING_TSTAMP, + SNDRV_RAWMIDI_FRAMING_LAST = SNDRV_RAWMIDI_FRAMING_TSTAMP, +}; + +#define SND_RAWMIDI_FRAMING_DATA_LENGTH 16 + +struct snd_rawmidi_framing_tstamp { + /* For now, frame_type is always 0. Midi 2.0 is expected to add new + * types here. Applications are expected to skip unknown frame types. + */ + unsigned char frame_type; + unsigned char length; /* number of valid bytes in data field */ + unsigned char reserved[2]; + unsigned int tv_nsec; /* nanoseconds */ + unsigned long long tv_sec; /* seconds */ + unsigned char data[SND_RAWMIDI_FRAMING_DATA_LENGTH]; +}; + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned char framing; /* For input data only, frame incoming data */ + unsigned char clock_type; /* Type of clock to use for framing, same as clockid_t */ + unsigned char reserved[14]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..d4b6b9b5c0e4 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -683,6 +683,8 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; + if (params->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP && params->buffer_size & 0x1f) + return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { @@ -720,7 +722,16 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { + if (params->framing) { + if (params->framing > SNDRV_RAWMIDI_FRAMING_LAST) + return -EINVAL; + /* framing requires a valid clock type */ + if (params->clock_type != CLOCK_MONOTONIC_RAW && params->clock_type != CLOCK_MONOTONIC) + return -EINVAL; + } snd_rawmidi_drain_input(substream); + substream->framing = params->framing; + substream->clock_type = params->clock_type; return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +974,42 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp *dest_ptr; + struct snd_rawmidi_framing_tstamp frame = { .tv_sec = tstamp->tv_sec, .tv_nsec = tstamp->tv_nsec }; + + int dest_frames = 0; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + if (snd_BUG_ON(runtime->hw_ptr & 0x1f || runtime->buffer_size & 0x1f || frame_size != 0x20)) + return -EINVAL; + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + return dest_frames * frame_size; + } + if (src_count >= SND_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SND_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SND_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return dest_frames * frame_size; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1024,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64; int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1035,15 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, "snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; } - spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP) { + if (substream->clock_type == CLOCK_MONOTONIC_RAW) + ktime_get_raw_ts64(&ts64); + else + ktime_get_ts64(&ts64); + spin_lock_irqsave(&runtime->lock, flags); + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ + spin_lock_irqsave(&runtime->lock, flags); substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -999,6 +1054,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, runtime->xruns++; } } else { + spin_lock_irqsave(&runtime->lock, flags); substream->bytes += count; count1 = runtime->buffer_size - runtime->hw_ptr; if (count1 > count) -- 2.25.1

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[PATCH v2 00/12] ASoC: simple-card-utils: prepare for multi support
by Kuninori Morimoto 13 Apr '21

13 Apr '21

Hi Mark This patch-set is for v2 of Multi-CPU/Codec support, and some cleanups. v1 had patch-conflict on simple-card / audio-graph with below. v2 was solved it. fa74c223b6fd78a5314b4c61b9abdbed3c2185b4 ("ASoC: simple-card: fix possible uninitialized single_cpu local variable") I want to add new audio-graph-card2 driver which can support not only DPCM, but also Multi-CPU/Codec, and Codec2Codec. And it is also supporting audio-graph-card2 base custom driver. But before supporting such driver, we need to cleanup existing simple-card / audio-graph, because these and new driver are sharing code. Link: https://lore.kernel.org/r/87wntmod33.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/20210408141639.GA39604@sirena.org.uk Kuninori Morimoto (12): ASoC: simple-card-utils: setup dai_props cpu_dai/codec_dai at initial timing ASoC: simple-card-utils: remove li->dais/li->conf ASoC: simple-card-utils: use for_each_prop_xxx() ASoC: simple-card-utils: remove asoc_simple_parse_xxx() ASoC: simple-card-utils: care multi DAI at asoc_simple_clean_reference() ASoC: simple-card-utils: indicate dai_fmt if exist ASoC: simple-card-utils: indicate missing CPU/Codec numbers for debug ASoC: simple-card-utils: add simple_props_to_xxx() macro ASoC: simple-card-utils: multi support at asoc_simple_canonicalize_cpu/platform() ASoC: simple-card-utils: tidyup debug info for clock ASoC: simple-card-utils: tidyup dev_dbg() to use 1 line ASoC: simple-card-utils: tidyup asoc_simple_parse_convert() include/sound/simple_card_utils.h | 107 ++++++++++------ sound/soc/generic/audio-graph-card.c | 64 ++++------ sound/soc/generic/simple-card-utils.c | 174 ++++++++++++++++---------- sound/soc/generic/simple-card.c | 70 ++++------- 4 files changed, 226 insertions(+), 189 deletions(-) -- 2.25.1

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[PATCH -next] ASoC: sti: sti_uniperif: add missing MODULE_DEVICE_TABLE
by Chen Lifu 13 Apr '21

13 Apr '21

This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Chen Lifu <chenlifu(a)huawei.com> --- sound/soc/sti/sti_uniperif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 67315d9b352d..e3561f00ed40 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -97,6 +97,7 @@ static const struct of_device_id snd_soc_sti_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, snd_soc_sti_match); int sti_uniperiph_reset(struct uniperif *uni) {

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[PATCH] ASoC: cs35l36: Fix an error handling path in 'cs35l36_i2c_probe()'
by Christophe JAILLET 13 Apr '21

13 Apr '21

If 'devm_regmap_init_i2c()' fails, there is no need to goto err. We should return directly as already done by the surrounding error handling paths. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/cs35l36.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index 4451ca9f4916..a038bcec2d17 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -1721,7 +1721,7 @@ static int cs35l36_i2c_probe(struct i2c_client *i2c_client, if (IS_ERR(cs35l36->regmap)) { ret = PTR_ERR(cs35l36->regmap); dev_err(dev, "regmap_init() failed: %d\n", ret); - goto err; + return ret; } cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); -- 2.27.0

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[PATCH] ASoC: meson: axg-frddr: fix fifo depth on g12 and sm1
by Jerome Brunet 13 Apr '21

13 Apr '21

Previous fifo depth patch was only tested on axg, not g12 or sm1. Of course, while adding hw_params dai callback for the axg, I forgot to do the same for g12 and sm1, leaving the depth unset and breaking playback on these SoCs. Add hw_params callback to the g12 dai_ops to fix the problem. Fixes: 6f68accaa864 ("ASoC: meson: axg-frddr: set fifo depth according to the period") Reported-by: Christian Hewitt <christianshewitt(a)gmail.com> Tested-by: Christian Hewitt <christianshewitt(a)gmail.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-frddr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index 8ed114de0bdf..37f4bb3469b5 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -171,6 +171,7 @@ static const struct axg_fifo_match_data axg_frddr_match_data = { static const struct snd_soc_dai_ops g12a_frddr_ops = { .prepare = g12a_frddr_dai_prepare, + .hw_params = axg_frddr_dai_hw_params, .startup = axg_frddr_dai_startup, .shutdown = axg_frddr_dai_shutdown, }; -- 2.30.2

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rockchip/veyron sound level issues
by GitHub issues - opened 13 Apr '21

13 Apr '21

alsa-project/alsa-ucm-conf issue #87 was opened from kapouer: Hi, setup: > asus c201 (speedy veyron) > linux 5.10.28 or 5.11.13 > debian bullseye: alsa 1.2.4, asa-ucm-conf (and topology) installed, pulseaudio 14.2, gnome 3.38 on wayland what works: - jack input/output using headset problems: - speaker output is somewhat low even at 100% (feels like 50%) - microphone is so low that it cannot detect sound (smashing the mic hole with a stick is detected but barely audible) - last time i checked hdmi was working - i need to check again though What can i provide ? pulseaudio -vvvvvv ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/87 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: ak5558: correct reset polarity
by Shengjiu Wang 13 Apr '21

13 Apr '21

Reset (aka power off) happens when the reset gpio is made active. Change function name to ak5558_reset to match devicetree property "reset-gpios". Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak5558.c | 30 ++++++++++-------------------- 1 file changed, 10 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 8e4dca753f0b..5c3f15827423 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -318,29 +318,19 @@ static struct snd_soc_dai_driver ak5552_dai = { .ops = &ak5558_dai_ops, }; -static void ak5558_power_off(struct ak5558_priv *ak5558) +static void ak5558_reset(struct ak5558_priv *ak5558, bool active) { - if (!ak5558->reset_gpiod) - return; - - gpiod_set_value_cansleep(ak5558->reset_gpiod, 0); - usleep_range(1000, 2000); -} - -static void ak5558_power_on(struct ak5558_priv *ak5558) -{ - if (!ak5558->reset_gpiod) - return; - - gpiod_set_value_cansleep(ak5558->reset_gpiod, 1); - usleep_range(1000, 2000); + if (ak5558->reset_gpiod) { + gpiod_set_value_cansleep(ak5558->reset_gpiod, active); + usleep_range(1000, 2000); + } } static int ak5558_probe(struct snd_soc_component *component) { struct ak5558_priv *ak5558 = snd_soc_component_get_drvdata(component); - ak5558_power_on(ak5558); + ak5558_reset(ak5558, false); return ak5558_set_mcki(component); } @@ -348,7 +338,7 @@ static void ak5558_remove(struct snd_soc_component *component) { struct ak5558_priv *ak5558 = snd_soc_component_get_drvdata(component); - ak5558_power_off(ak5558); + ak5558_reset(ak5558, true); } static int __maybe_unused ak5558_runtime_suspend(struct device *dev) @@ -356,7 +346,7 @@ static int __maybe_unused ak5558_runtime_suspend(struct device *dev) struct ak5558_priv *ak5558 = dev_get_drvdata(dev); regcache_cache_only(ak5558->regmap, true); - ak5558_power_off(ak5558); + ak5558_reset(ak5558, true); regulator_bulk_disable(ARRAY_SIZE(ak5558->supplies), ak5558->supplies); @@ -375,8 +365,8 @@ static int __maybe_unused ak5558_runtime_resume(struct device *dev) return ret; } - ak5558_power_off(ak5558); - ak5558_power_on(ak5558); + ak5558_reset(ak5558, true); + ak5558_reset(ak5558, false); regcache_cache_only(ak5558->regmap, false); regcache_mark_dirty(ak5558->regmap); -- 2.27.0

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[PATCH 0/5] ASoC: clock provider clean-up
by Jerome Brunet 13 Apr '21

13 Apr '21

The purpose of this patchset it remove the use the clk member of 'struct clk_hw' in ASoC. 'struct clk' is a per-user reference to an actual clock. In the future, the clk member in 'struct clk_hw' may go away. The usage of this member by a clock provider usually falls into either of following categories: * Mis-usage of the clock consumer API by a clock provider. * Clock provider also being a user of its own clocks. In this case the provider should request a 'struct clk' through the appropriate API instead of poking in 'struct clk_hw' internals. Jerome Brunet (5): ASoC: stm32: properly get clk from the provider ASoC: wcd934x: use the clock provider API ASoC: rt5682: clock driver must use the clock provider API ASoC: lpass: use the clock provider API ASoC: da7219: properly get clk from the provider sound/soc/codecs/da7219.c | 5 ++++- sound/soc/codecs/lpass-va-macro.c | 2 +- sound/soc/codecs/lpass-wsa-macro.c | 9 +++------ sound/soc/codecs/rt5682.c | 6 +++--- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/stm/stm32_sai_sub.c | 5 ++++- 6 files changed, 19 insertions(+), 14 deletions(-) -- 2.30.2

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[PATCH v2] Behringer UFX1604 / UFX1204: get rid of unneeded implicit feedback and pops and clicks while on 96000hz
by Geraldo Nascimento 13 Apr '21

13 Apr '21

More complete patch disabling unneeded implicit feedback and setting clock selector to default clock on rate change for UFX1604 After re-reading https://bugzilla.kernel.org/show_bug.cgi?id=199327 it is even more clear to me that implicit feedback for the UFX1604/UFX1204 needs to be disabled. This is a more complete patch that disables that and for the UFX1604 only sets the clock selector to its pin 1 default clock synced to the USB SOF upon rate change. This is needed because apparently the endpoints are hardwired to the clock selector and after we change the rate on the main USB SOF synced clock the clock selector is left in a halfway state in regards to the sampling rate. That's why the pops and clicks aren't evident at stock 48000Hz, become slightly audible at 44100Hz and detestable at 96000Hz. Seems the clock selector needs a nudge or it will screw up the sync. Unfortunately I don't have access to the lsusb -v of the UFX1204 soI'm waiting for someone to share it here in the list or in the bugzilla thread. This patch needs some more love from the community. --- This one has been bugging me for quite a while. I went deep hard in the guts of ALSA to try to solve it, and it turned out to be a minor thing apparently. The problem is old, and every UFX1604 Linux user can attest that it's impossible to use 96000hz in DUPLEX mode without annoying pops and clicks on the capture stream. The fix is simple: after we alter the CLOCK_SOURCE to match our sample rate, let's tell the CLOCK_SELECTOR we want CLOCK_SOURCE 212 (synced to USB SOF) on pin 1. Solves the problem for me, no more pops and clicks while on 96000hz. --- Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- implicit.c 2021-04-04 20:51:57.226754632 -0300 +++ implicit.c 2021-04-09 21:55:19.593994214 -0300 @@ -50,8 +50,6 @@ static const struct snd_usb_implicit_fb_ /* Fixed EP */ /* FIXME: check the availability of generic matching */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0001, 0x81, 1), /* Behringer UFX1604 */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0002, 0x81, 1), /* Behringer UFX1204 */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8010, 0x81, 2), /* Fractal Audio Axe-Fx III */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0001, 0x81, 2), /* Solid State Logic SSL2 */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0002, 0x81, 2), /* Solid State Logic SSL2+ */ --- clock.c 2021-04-09 22:01:08.234001611 -0300 +++ clock.c 2021-04-09 22:01:53.640669243 -0300 @@ -610,6 +610,14 @@ int snd_usb_set_sample_rate_v2v3(struct if (err < 0) return err; + if (chip->usb_id == USB_ID(0x1397, 0x0001)) { /* Behringer UFX1604 */ + printk(KERN_WARNING "Setting clock selector for UFX1604"); + err = uac_clock_selector_set_val(chip, 211, 1); + if (err < 0) + return err; + } + + return get_sample_rate_v2v3(chip, fmt->iface, fmt->altsetting, clock); }

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[PATCH 0/2] ASoC: SOF: Intel: descriptor corrections for TGL and ADL
by Pierre-Louis Bossart 12 Apr '21

12 Apr '21

One missed property for TigerLake and need for separate descriptors between ADL-S and the other flavors. Libin Yang (1): ASoC: SOF: Intel: add missing use_acpi_target_states for TGL platforms Sathya Prakash M R (1): ASoC: SOF: Intel: Update ADL P to use its own descriptor sound/soc/sof/intel/pci-tgl.c | 20 +++++++++++++++++++- sound/soc/sof/intel/tgl.c | 2 +- 2 files changed, 20 insertions(+), 2 deletions(-) -- 2.25.1

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