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

[RFC PATCH 0/3] Separate BE DAI HW constraints from FE ones
by Codrin Ciubotariu 19 Apr '21

19 Apr '21

HW constraints are needed to set limitations for HW parameters used to configure the DAIs. All DAIs on the same link must agree upon the HW parameters, so the parameters are affected by the DAIs' features and their limitations. In case of DPCM, the FE DAIs can be used to perform different kind of conversions, such as format or rate changing, bringing the audio stream to a configuration supported by all the DAIs of the BE's link. For this reason, the limitations of the BE DAIs are not always important for the HW parameters between user-space and FE, only for the paratemers between FE and BE DAI links. This brings us to this patch-set, which aims to separate the FE HW constraints from the BE HW constraints. This way, the FE can be used to perform more efficient HW conversions, on the initial audio stream parameters, to parameters supported by the BE DAIs. To achieve this, the first thing needed is to detect whether a HW constraint rule is enforced by a FE or a BE DAI. This means that snd_pcm_hw_rule_add() needs to be able to differentiate between the two type of DAIs. For this, the runtime pointer to struct snd_pcm_runtime is replaced with a pointer to struct snd_pcm_substream, to be able to reach substream->pcm->internal to differentiate between FE and BE DAIs. This change affects many sound drivers (and one gpu drm driver). All these changes are included in the first patch, to have a better overview of the implications created by this change. The second patch adds a new struct snd_pcm_hw_constraints under struct snd_soc_dpcm_runtime, which is used to store the HW constraint rules added by the BE DAIs. This structure is initialized with a subset of the runtime constraint rules which does not include the rules that affect the buffer or period size. snd_pcm_hw_rule_add() will add the BE rules to the new struct snd_pcm_hw_constraints. The third and last patch will apply the BE rule constraints, after the fixup callback. If the fixup HW parameters do not respect the BE constraint rules, the rules will exit with an error. The FE mask and interval constraints are copied to the BE ones, to satisfy the dai_link->dpcm_merged_* flags. The dai_link->dpcm_merged_* flags are used to know if the FE does format or sampling rate conversion. I tested with ad1934 and wm8731 codecs as BEs, with a not-yet-mainlined ASRC as FE, that can also do format conversion. I realize that the change to snd_pcm_hw_rule_add() has a big impact, even though all the drivers use snd_pcm_hw_rule_add() with substream->runtime, so passing substream instead of runtime is not that risky. Codrin Ciubotariu (3): pcm: use substream instead of runtime in snd_pcm_hw_rule_add() ASoC: soc-pcm: add hw_constraints for BE DAI links ASoC: soc-pcm: apply BE HW constraint rules drivers/gpu/drm/vc4/vc4_hdmi.c | 2 +- include/sound/ac97_codec.h | 2 +- include/sound/pcm.h | 29 +++++-- include/sound/pcm_drm_eld.h | 2 +- include/sound/soc-dpcm.h | 1 + sound/arm/aaci.c | 4 +- sound/arm/pxa2xx-pcm-lib.c | 4 +- sound/core/pcm_drm_eld.c | 6 +- sound/core/pcm_lib.c | 66 ++++++++------- sound/core/pcm_native.c | 83 +++++++++++-------- sound/drivers/aloop.c | 8 +- sound/drivers/vx/vx_pcm.c | 8 +- sound/firewire/amdtp-am824.c | 8 +- sound/firewire/amdtp-am824.h | 2 +- sound/firewire/amdtp-stream.c | 12 +-- sound/firewire/amdtp-stream.h | 2 +- sound/firewire/bebob/bebob_pcm.c | 6 +- sound/firewire/dice/dice-pcm.c | 6 +- sound/firewire/digi00x/amdtp-dot.c | 6 +- sound/firewire/digi00x/digi00x-pcm.c | 6 +- sound/firewire/digi00x/digi00x.h | 2 +- sound/firewire/fireface/amdtp-ff.c | 6 +- sound/firewire/fireface/ff-pcm.c | 6 +- sound/firewire/fireface/ff.h | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 6 +- sound/firewire/motu/amdtp-motu.c | 6 +- sound/firewire/motu/motu-pcm.c | 6 +- sound/firewire/motu/motu.h | 2 +- sound/firewire/oxfw/oxfw-pcm.c | 6 +- sound/firewire/tascam/amdtp-tascam.c | 6 +- sound/firewire/tascam/tascam-pcm.c | 2 +- sound/firewire/tascam/tascam.h | 2 +- sound/pci/ac97/ac97_pcm.c | 8 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 6 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 8 +- sound/pci/ca0106/ca0106_main.c | 4 +- sound/pci/cmipci.c | 14 ++-- sound/pci/cs4281.c | 4 +- sound/pci/cs46xx/cs46xx_lib.c | 4 +- sound/pci/echoaudio/echoaudio.c | 24 +++--- sound/pci/emu10k1/emu10k1x.c | 4 +- sound/pci/emu10k1/emupcm.c | 10 +-- sound/pci/ens1370.c | 12 +-- sound/pci/es1938.c | 4 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 6 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.c | 4 +- sound/pci/hda/patch_hdmi.c | 8 +- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/ice1712/ice1712.c | 8 +- sound/pci/ice1712/ice1724.c | 30 +++---- sound/pci/intel8x0.c | 10 +-- sound/pci/intel8x0m.c | 2 +- sound/pci/lola/lola_pcm.c | 4 +- sound/pci/mixart/mixart.c | 8 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/oxygen_pcm.c | 8 +- sound/pci/pcxhr/pcxhr.c | 4 +- sound/pci/rme32.c | 14 ++-- sound/pci/rme96.c | 17 ++-- sound/pci/rme9652/hdsp.c | 24 +++--- sound/pci/rme9652/hdspm.c | 12 +-- sound/pci/rme9652/rme9652.c | 20 ++--- sound/pci/sonicvibes.c | 4 +- sound/pci/via82xx.c | 4 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 4 +- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp-da7219-max98357a.c | 40 ++++----- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 4 +- sound/soc/bcm/cygnus-pcm.c | 5 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/adau1372.c | 2 +- sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak4613.c | 6 +- sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 4 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs4234.c | 4 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/cs43130.c | 4 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/pcm512x.c | 6 +- sound/soc/codecs/sigmadsp.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/uda1334.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8524.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/fsl/fsl_asrc.c | 4 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_sai.c | 4 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 9 +- sound/soc/fsl/imx-audmix.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 12 +-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 12 +-- sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/aiu-fifo.c | 4 +- sound/soc/meson/axg-fifo.c | 4 +- sound/soc/sh/rcar/core.c | 6 +- sound/soc/soc-pcm.c | 76 ++++++++++++++++- sound/soc/sti/uniperif_player.c | 4 +- sound/soc/sti/uniperif_reader.c | 4 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra_pcm.c | 2 +- sound/soc/ti/davinci-mcasp.c | 12 +-- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/uniphier/aio-dma.c | 4 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- sound/usb/hiface/pcm.c | 2 +- sound/usb/line6/capture.c | 2 +- sound/usb/line6/playback.c | 2 +- sound/usb/misc/ua101.c | 2 +- sound/usb/pcm.c | 24 +++--- 135 files changed, 535 insertions(+), 429 deletions(-) -- 2.27.0

5 20

[PATCH] ALSA: seq: oss: Fix inconsistent indenting
by Yang Li 19 Apr '21

19 Apr '21

Kernel test robot throws below warning -> smatch warnings: sound/core/seq/oss/seq_oss_event.c:297 note_on_event() warn: inconsistent indenting Fixed the inconsistent indenting. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/core/seq/oss/seq_oss_event.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c index 7b7c925..9a42713 100644 --- a/sound/core/seq/oss/seq_oss_event.c +++ b/sound/core/seq/oss/seq_oss_event.c @@ -294,7 +294,7 @@ /* set volume to zero -- note off */ // type = SNDRV_SEQ_EVENT_NOTEOFF; //else - if (info->ch[ch].vel) + if (info->ch[ch].vel) /* sample already started -- volume change */ type = SNDRV_SEQ_EVENT_KEYPRESS; else -- 1.8.3.1

3 2

[RESEND PATCH] pinctrl: mediatek: use spin lock in mtk_rmw
by Tzung-Bi Shih 19 Apr '21

19 Apr '21

Commit 42a46434e9b1 ("pinctrl: add lock in mtk_rmw function.") uses mutex lock in mtk_rmw. However the function is possible called from atomic context. For example call trace: mutex_lock+0x28/0x64 mtk_rmw+0x38/0x80 mtk_hw_set_value+0x100/0x138 mtk_gpio_set+0x48/0x58 gpiod_set_raw_value_commit+0xf4/0x110 gpiod_set_value_nocheck+0x4c/0x80 gpiod_set_value+0x4c/0x6c max98357a_daiops_trigger+0x8c/0x9c soc_pcm_trigger+0x5c/0x10c The max98357a_daiops_trigger() could run in either atomic or non-atomic context. As a result, dmesg shows some similar messages: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:254". Uses spin lock in mtk_rmw instead. Fixes: 42a46434e9b1 ("pinctrl: add lock in mtk_rmw function.") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- drivers/pinctrl/mediatek/pinctrl-moore.c | 2 +- drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c | 4 ++-- drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h | 2 +- drivers/pinctrl/mediatek/pinctrl-paris.c | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/pinctrl/mediatek/pinctrl-moore.c b/drivers/pinctrl/mediatek/pinctrl-moore.c index f77921957f15..3a4a23c40a71 100644 --- a/drivers/pinctrl/mediatek/pinctrl-moore.c +++ b/drivers/pinctrl/mediatek/pinctrl-moore.c @@ -619,7 +619,7 @@ int mtk_moore_pinctrl_probe(struct platform_device *pdev, hw->nbase = hw->soc->nbase_names; - mutex_init(&hw->lock); + spin_lock_init(&hw->lock); /* Copy from internal struct mtk_pin_desc to register to the core */ pins = devm_kmalloc_array(&pdev->dev, hw->soc->npins, sizeof(*pins), diff --git a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c index fcf7c3eeee4a..9ffe01a8ceca 100644 --- a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c +++ b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c @@ -58,14 +58,14 @@ void mtk_rmw(struct mtk_pinctrl *pctl, u8 i, u32 reg, u32 mask, u32 set) { u32 val; - mutex_lock(&pctl->lock); + spin_lock(&pctl->lock); val = mtk_r32(pctl, i, reg); val &= ~mask; val |= set; mtk_w32(pctl, i, reg, val); - mutex_unlock(&pctl->lock); + spin_unlock(&pctl->lock); } static int mtk_hw_pin_field_lookup(struct mtk_pinctrl *hw, diff --git a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h index 65eac708a3b3..13836bb188b7 100644 --- a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h +++ b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h @@ -252,7 +252,7 @@ struct mtk_pinctrl { struct mtk_pinctrl_group *groups; const char **grp_names; /* lock pin's register resource to avoid multiple threads issue*/ - struct mutex lock; + spinlock_t lock; }; void mtk_rmw(struct mtk_pinctrl *pctl, u8 i, u32 reg, u32 mask, u32 set); diff --git a/drivers/pinctrl/mediatek/pinctrl-paris.c b/drivers/pinctrl/mediatek/pinctrl-paris.c index 48e823f6d293..85db2e4377f0 100644 --- a/drivers/pinctrl/mediatek/pinctrl-paris.c +++ b/drivers/pinctrl/mediatek/pinctrl-paris.c @@ -970,7 +970,7 @@ int mtk_paris_pinctrl_probe(struct platform_device *pdev, hw->nbase = hw->soc->nbase_names; - mutex_init(&hw->lock); + spin_lock_init(&hw->lock); err = mtk_pctrl_build_state(pdev); if (err) { -- 2.31.1.368.gbe11c130af-goog

2 2

[PATCH v5] sound: rawmidi: Add framing mode
by David Henningsson 19 Apr '21

19 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> --- This version of the patch has been compile tested only, and compared to v4, has only the changes with no controversy. include/sound/rawmidi.h | 2 + include/uapi/sound/asound.h | 26 ++++++++++++- sound/core/rawmidi.c | 73 ++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 6 ++- 4 files changed, 103 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..124ac74a13e9 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. + */ + u8 frame_type; + u8 length; /* number of valid bytes in data field */ + u8 reserved[2]; + u32 tv_nsec; /* nanoseconds */ + u64 tv_sec; /* seconds */ + u8 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..c3a4940a919d 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) != 0) + 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 +974,56 @@ 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, const 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); + + BUILD_BUG_ON(frame_size != 0x20); + if (snd_BUG_ON((runtime->hw_ptr & 0x1f) != 0)) + return -EINVAL; + + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + break; + } + 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; +} + +struct timespec64 get_framing_tstamp(struct snd_rawmidi_substream *substream) +{ + struct timespec64 ts64 = {0, 0}; + + if (substream->framing != SNDRV_RAWMIDI_FRAMING_TSTAMP) + return ts64; + if (substream->clock_type == CLOCK_MONOTONIC_RAW) + ktime_get_raw_ts64(&ts64); + else + ktime_get_ts64(&ts64); + return ts64; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1038,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64 = get_framing_tstamp(substream); int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1049,11 @@ 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) { + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -1596,6 +1661,12 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Avail : %lu\n" " Overruns : %lu\n", buffer_size, avail, xruns); + if (substream->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP) { + snd_iprintf(buffer, + " Framing : tstamp\n" + " Clock type : %s\n", + substream->clock_type == CLOCK_MONOTONIC_RAW ? "monotonic raw" : "monotonic"); + } } } } diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c index 7397130976d0..2603d2dd8abb 100644 --- a/sound/core/rawmidi_compat.c +++ b/sound/core/rawmidi_compat.c @@ -13,7 +13,9 @@ struct snd_rawmidi_params32 { u32 buffer_size; u32 avail_min; unsigned int no_active_sensing; /* avoid bit-field */ - unsigned char reserved[16]; + unsigned char framing; + unsigned char clock_type; + unsigned char reserved[14]; } __attribute__((packed)); static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, @@ -25,6 +27,8 @@ static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, if (get_user(params.stream, &src->stream) || get_user(params.buffer_size, &src->buffer_size) || get_user(params.avail_min, &src->avail_min) || + get_user(params.framing, &src->framing) || + get_user(params.clock_type, &src->clock_type) || get_user(val, &src->no_active_sensing)) return -EFAULT; params.no_active_sensing = val; -- 2.25.1

4 7

[PATCH] ALSA: hda/realtek: GA503 use same quirks as GA401
by Luke D Jones 19 Apr '21

19 Apr '21

The GA503 has almost exactly the same default setup as the GA401 model with the same issues. The GA401 quirks solve all the issues so we will use the full quirk chain. Signed-off-by: Luke D Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..9a5546df1e6a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8019,6 +8019,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), + SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), -- 2.31.1

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[PATCH] ALSA: usb-audio: DJM-750: ensure format is set
by Olivia Mackintosh 19 Apr '21

19 Apr '21

Add case statement to set sample-rate for the DJM-750 Pioneer mixer. This was included as part of another patch but I think it has been archived on Patchwork and hasn't been merged. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 63f27ce64bcb..8b8bee3c3dd6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1511,6 +1511,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ pioneer_djm_set_format_quirk(subs, 0x0082); break; + case USB_ID(0x08e4, 0x017f): /* Pioneer DJM-750 */ case USB_ID(0x08e4, 0x0163): /* Pioneer DJM-850 */ pioneer_djm_set_format_quirk(subs, 0x0086); break; -- 2.31.1

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[PATCH v2] Pioneer devices: engage implicit feedback sync for playback
by Geraldo 19 Apr '21

19 Apr '21

Dear Linux users of Pioneer gear, please disregard v1 of this patch and test this instead if at all possible. My initial assessment that we needed to let the capture EP be opened twice was clearly proven wrong by further testing. This is good because if we really needed that it would require a lot of reengineering inside ALSA. One thing that still boggles my mind though is why we need a special Pioneer case inside snd_usb_parse_implicit_fb_quirk that returns 1 like a capture quirk was found. This is a highly experimental patch on top of 5.12-rc6 that's supposed to engage implicit feedback sync on the playback for Pioneer devices. Without implicit feedback sync the inputs started glitching due to clock drift in about an hour in my Pioneer DJ DDJ-SR2. Once again I'd like to thank Takashi Iwai. He's the true author of the bulk of this code, I just adapted it and coerced it into working. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- implicit.c.git 2021-04-04 20:51:57.226754632 -0300 +++ implicit.c 2021-04-05 10:02:41.059816581 -0300 @@ -177,30 +177,31 @@ static int add_roland_implicit_fb(struct ifnum, alts); } -/* Playback and capture EPs on Pioneer devices share the same iface/altset, - * but they don't seem working with the implicit fb mode well, hence we - * just return as if the sync were already set up. +/* Pioneer devices: playback and capture streams sharing the same iface/altset */ -static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, - struct audioformat *fmt, - struct usb_host_interface *alts) -{ - struct usb_endpoint_descriptor *epd; - - if (alts->desc.bNumEndpoints != 2) - return 0; - - epd = get_endpoint(alts, 1); - if (!usb_endpoint_is_isoc_in(epd) || - (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC || - ((epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != - USB_ENDPOINT_USAGE_DATA && - (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != - USB_ENDPOINT_USAGE_IMPLICIT_FB)) - return 0; - return 1; /* don't handle with the implicit fb, just skip sync EP */ +static int add_pioneer_implicit_fb(struct snd_usb_audio *chip, + struct audioformat *fmt, + struct usb_host_interface *alts) +{ + struct usb_endpoint_descriptor *epd; + + if (alts->desc.bNumEndpoints != 2) + return 0; + + epd = get_endpoint(alts, 1); + + if (!usb_endpoint_is_isoc_in(epd) || + (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC || + ((epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != + USB_ENDPOINT_USAGE_DATA && + (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != + USB_ENDPOINT_USAGE_IMPLICIT_FB)) + return 0; + return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 1, + alts->desc.bInterfaceNumber, alts); } + static int __add_generic_implicit_fb(struct snd_usb_audio *chip, struct audioformat *fmt, int iface, int altset) @@ -306,7 +307,7 @@ static int audioformat_implicit_fb_quirk alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) { - if (skip_pioneer_sync_ep(chip, fmt, alts)) + if (add_pioneer_implicit_fb(chip, fmt, alts)) return 1; } @@ -339,8 +340,20 @@ int snd_usb_parse_implicit_fb_quirk(stru struct audioformat *fmt, struct usb_host_interface *alts) { - if (fmt->endpoint & USB_DIR_IN) - return audioformat_capture_quirk(chip, fmt, alts); + bool isPioneer; + + if (alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && + (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ + USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) + isPioneer = true; + + if (fmt->endpoint & USB_DIR_IN) { + if (isPioneer == true) + return 1; + else + return audioformat_capture_quirk(chip, fmt, alts); + } + else return audioformat_implicit_fb_quirk(chip, fmt, alts); }

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[RFC] ucm2: rt711-sdca: remove 'rt711 FU0F Capture Switch' setting
by GitHub pull_request - edited 19 Apr '21

19 Apr '21

alsa-project/alsa-ucm-conf pull request #89 was edited from libinyang: The rt711-sdca codec driver removes the 'rt711 FU0F Capture Switch' kcontrol. So we need to remove the setting of 'rt711 FU0F Capture Switch' correspondingly. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/89 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/89.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: rt711-sdca: remove 'rt711 FU0F Capture Switch' setting
by GitHub pull_request - opened 19 Apr '21

19 Apr '21

alsa-project/alsa-ucm-conf pull request #89 was opened from libinyang: The rt711-sdca codec driver removes the 'rt711 FU0F Capture Switch' kcontrol. So we need to remove the setting of 'rt711 FU0F Capture Switch' correspondingly. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/89 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/89.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/4] soundwire: only use CLOCK_STOP_MODE0 and handle -ENODATA
by Bard Liao 19 Apr '21

19 Apr '21

Existing devices and implementations only support the required CLOCK_STOP_MODE0. All the code related to CLOCK_STOP_MODE1 has not been tested and is highly questionable, with a clear confusion between CLOCK_STOP_MODE1 and the simple clock stop state machine. This patch removes all usages of CLOCK_STOP_MODE1 - which has no impact on any solution - and fixes the use of the simple clock stop state machine. The resulting code should be a lot more symmetrical and easier to maintain. Note that CLOCK_STOP_MODE1 is not supported in the SoundWire Device Class specification so it's rather unlikely that we need to re-add this mode later. If a device lost sync and can no longer ACK a command, it may not be able to enter a lower-power state but it will still be able to resync when the clock restarts. In those cases, we want to continue with the clock stop sequence. This patch modifies the behavior during clock stop sequences to only log errors unrelated to -ENODATA/Command_Ignored. The flow is also modified so that loops continue to prepare/deprepare other devices even when one seems to have lost sync. When resuming the clocks, all issues are logged with a dev_warn(), previously only some of them were checked. This is the only part that now differs between the clock stop entry and clock stop exit sequences: while we don't want to stop the suspend flow, we do want information on potential issues while resuming, as they may have ripple effects. For consistency the log messages are also modified to be unique and self-explanatory. Errors in sdw_slave_clk_stop_callback() were removed, they are now handled in the caller. Pierre-Louis Bossart (4): soundwire: bus: only use CLOCK_STOP_MODE0 and fix confusions soundwire: add missing kernel-doc description soundwire: bus: handle -ENODATA errors in clock stop/start sequences soundwire: bus: add missing \n in dynamic debug drivers/soundwire/bus.c | 155 +++++++++++++++------------------- include/linux/soundwire/sdw.h | 3 +- 2 files changed, 70 insertions(+), 88 deletions(-) -- 2.17.1

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