September 2023 - Alsa-devel - mailman.alsa-project.org

Update alc287_fixup_bind_dacs()
by Kailang 12 Sep '23

12 Sep '23

Hi Takashi, Two speakers platform will common use same function. So, attach patch was an update. BR, Kailang

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[PATCH] ALSA: seq: Avoid delivery of events for disabled UMP groups
by Takashi Iwai 12 Sep '23

12 Sep '23

ALSA sequencer core still delivers events to the disabled UMP Group, leaving this handling to the device. But it's rather risky and it's easy to imagine that such an unexpected event may screw up the device firmware. This patch avoids the superfluous event deliveries by setting the group_filter of the UMP client as default, and evaluate the group_filter properly at delivery from non-UMP clients. The grouop_filter is updated upon the dynamic UMP Function Block updates, so that it follows the change of the disabled UMP Groups, too. Fixes: d2b706077792 ("ALSA: seq: Add UMP group filter") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_ump_client.c | 22 ++++++++++++++++++++++ sound/core/seq/seq_ump_convert.c | 2 ++ 2 files changed, 24 insertions(+) diff --git a/sound/core/seq/seq_ump_client.c b/sound/core/seq/seq_ump_client.c index f26a1812dfa7..a60e3f069a80 100644 --- a/sound/core/seq/seq_ump_client.c +++ b/sound/core/seq/seq_ump_client.c @@ -416,6 +416,25 @@ static void setup_client_midi_version(struct seq_ump_client *client) snd_seq_kernel_client_put(cptr); } +/* set up client's group_filter bitmap */ +static void setup_client_group_filter(struct seq_ump_client *client) +{ + struct snd_seq_client *cptr; + unsigned int filter; + int p; + + cptr = snd_seq_kernel_client_get(client->seq_client); + if (!cptr) + return; + filter = ~(1U << 0); /* always allow groupless messages */ + for (p = 0; p < SNDRV_UMP_MAX_GROUPS; p++) { + if (client->groups[p].active) + filter &= ~(1U << (p + 1)); + } + cptr->group_filter = filter; + snd_seq_kernel_client_put(cptr); +} + /* UMP group change notification */ static void handle_group_notify(struct work_struct *work) { @@ -424,6 +443,7 @@ static void handle_group_notify(struct work_struct *work) update_group_attrs(client); update_port_infos(client); + setup_client_group_filter(client); } /* UMP FB change notification */ @@ -492,6 +512,8 @@ static int snd_seq_ump_probe(struct device *_dev) goto error; } + setup_client_group_filter(client); + err = create_ump_endpoint_port(client); if (err < 0) goto error; diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index 7cc84e137999..b141024830ec 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -1197,6 +1197,8 @@ int snd_seq_deliver_to_ump(struct snd_seq_client *source, struct snd_seq_event *event, int atomic, int hop) { + if (dest->group_filter & (1U << dest_port->ump_group)) + return 0; /* group filtered - skip the event */ if (event->type == SNDRV_SEQ_EVENT_SYSEX) return cvt_sysex_to_ump(dest, dest_port, event, atomic, hop); else if (snd_seq_client_is_midi2(dest)) -- 2.35.3

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[PATCH] ALSA: docs: Fix a typo of midi2_ump_probe option for snd-usb-audio
by Takashi Iwai 12 Sep '23

12 Sep '23

A simple typo fix: midi2_probe => midi2_ump_probe. Fixes: febdfa0e9c8a ("ALSA: docs: Update MIDI 2.0 documentation for UMP 1.1 enhancement") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/designs/midi-2.0.rst | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/designs/midi-2.0.rst b/Documentation/sound/designs/midi-2.0.rst index 45987f256b97..086487ca7ab1 100644 --- a/Documentation/sound/designs/midi-2.0.rst +++ b/Documentation/sound/designs/midi-2.0.rst @@ -74,8 +74,8 @@ topology based on those information. When the device is older and doesn't respond to the new UMP inquiries, the driver falls back and builds the topology based on Group Terminal Block (GTB) information from the USB descriptor. Some device might be screwed up by the -unexpected UMP command; in such a case, pass `midi2_probe=0` option to -snd-usb-audio driver for skipping the UMP v1.1 inquiries. +unexpected UMP command; in such a case, pass `midi2_ump_probe=0` +option to snd-usb-audio driver for skipping the UMP v1.1 inquiries. When the MIDI 2.0 device is probed, the kernel creates a rawmidi device for each UMP Endpoint of the device. Its device name is -- 2.35.3

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RE: Re: [PATCH] ASoC: dwc: fix typo in comment
by 김명욱 12 Sep '23

12 Sep '23

ISR_RXFO means \"Status of Data Overrun interrupt for the RX channel\" according to the datasheetSo, the comment should be RX, not TXThanksMyunguk------------Original Message------------Subject : Re: [PATCH] ASoC: dwc: fix typo in commentDate : 2023-09-11 21:46:05From : Bagas Sanjaya <bagasdotme@gmailcom>To : Myunguk Kim <mwkim@gaonchipscom>, alsa-devel@alsa-projectorgCc : Liam Girdwood <lgirdwood@gmailcom>, Mark Brown <broonie@kernelorg>, Jaroslav Kysela <perex@perexcz>, Takashi Iwai <tiwai@susecom>, Maxim Kochetkov <fido_max@inboxru>, Kuninori Morimoto <kuninorimorimotogx@renesascom>, Xingyu Wu <xingyuwu@starfivetechcom>, Uwe Kleine-König <ukleine-koenig@pengutronixde>, linux-kernel@vgerkernelorgOn Mon, Sep 11, 2023 at 03:46:46PM +0900, Myunguk Kim wrote:> This fixes typo in commentOn what? Can you describe the typofix in more detail?>> Signed-off-by: Myunguk Kim <mwkim@gaonchipscom>> ---> sound/soc/dwc/dwc-i2sc | 2 +-> 1 file changed, 1 insertion(+), 1 deletion(-)>> diff --git a/sound/soc/dwc/dwc-i2sc b/sound/soc/dwc/dwc-i2sc> index 71ff894035a445e55e550924 100644> --- a/sound/soc/dwc/dwc-i2sc> +++ b/sound/soc/dwc/dwc-i2sc> @@ -138,7 +138,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id)> irq_valid = true;> }>> - /* Error Handling: TX */> + /* Error Handling: RX */> if (isr[i] & ISR_RXFO)> { dev_err_ratelimited(dev->dev, \"RX overrun (ch_id=%d)\\n\", i);> irq_valid = true;Oh, you mean the code following the comment checks for RXFO overrunConfused--An old man doll just what I always wanted! - Clara

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[PATCH] ASoC: fsl: imx-pcm-rpmsg: Add SNDRV_PCM_INFO_BATCH flag
by Shengjiu Wang 12 Sep '23

12 Sep '23

The rpmsg pcm device is a device which should support double buffering. Found this issue with pipewire. When there is no SNDRV_PCM_INFO_BATCH flag in driver, the pipewire will set headroom to be zero, and because rpmsg pcm device don't support residue report, when the latency setting is small, the "delay" always larger than "target" in alsa-pcm.c, that reading next period data is not scheduled on time. With SNDRV_PCM_INFO_BATCH flag in driver, the pipewire will select a smaller period size for device, then the task of reading next period data will be scheduled on time. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-pcm-rpmsg.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index d63782b8bdef..bb736d45c9e0 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -19,6 +19,7 @@ static struct snd_pcm_hardware imx_rpmsg_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | -- 2.34.1

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[PATCH v2 0/2] Fix redundant PLLA update
by Sameer Pujar 12 Sep '23

12 Sep '23

This small series fixes redundant PLLA updates that happen for each DAI link in the audio path. This helps to resolve DMIC clock issue seen on Jetson TX2 platform. Sameer Pujar (2): ASoC: soc-utils: Export snd_soc_dai_is_dummy() symbol ASoC: tegra: Fix redundant PLLA and PLLA_OUT0 updates sound/soc/soc-utils.c | 1 + sound/soc/tegra/tegra_audio_graph_card.c | 30 ++++++++++++++---------- 2 files changed, 18 insertions(+), 13 deletions(-) -- 2.17.1

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[PATCH v2 00/11] Improve CS35l41-based audio codec drivers
by Cristian Ciocaltea 12 Sep '23

12 Sep '23

This patch series contains several fixes and improvements to drivers based on the CS35l41 audio codec. It has been verified on Valve's Steam Deck, except the HDA related patches. Changes in v2: - Added Acked-by tags from Charles - Reworked the shared boost handling per reviews from Charles and David - Dropped PATCH 5/9 and moved PATCH 6/9 a bit down the list, as it doesn't contain a Fixes tag - Reverted usage of devm_pm_runtime_enable() in PATCH 9/9 and replaced with a proper runtime PM changes undo in PATCH v2 06/11 - Added 3 new patches providing some HDA related fixes - v1: https://lore.kernel.org/all/20230902210621.1184693-1-cristian.ciocaltea@col… Cristian Ciocaltea (11): ASoC: cs35l41: Handle mdsync_down reg write errors ASoC: cs35l41: Handle mdsync_up reg write errors ASoC: cs35l41: Initialize completion object before requesting IRQ ASoC: cs35l41: Fix broken shared boost activation ASoC: cs35l41: Verify PM runtime resume errors in IRQ handler ASoC: cs35l41: Undo runtime PM changes at driver exit time ASoC: cs35l41: Make use of dev_err_probe() ASoC: cs35l41: Use modern pm_ops ALSA: hda: cs35l41: Fix unbalanced pm_runtime_get() ALSA: hda: cs35l41: Undo runtime PM changes at driver exit time ALSA: hda: cs35l41: Consistently use dev_err_probe() include/sound/cs35l41.h | 4 +- sound/pci/hda/cs35l41_hda.c | 23 +++++---- sound/soc/codecs/cs35l41-i2c.c | 11 ++-- sound/soc/codecs/cs35l41-lib.c | 60 +++++++++++++--------- sound/soc/codecs/cs35l41-spi.c | 11 ++-- sound/soc/codecs/cs35l41.c | 93 +++++++++++++++++++--------------- sound/soc/codecs/cs35l41.h | 1 - 7 files changed, 112 insertions(+), 91 deletions(-) -- 2.41.0

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[PATCH v2 0/7] ASoC: Convert some Maxim codecs to use GPIO descriptors
by Linus Walleij 12 Sep '23

12 Sep '23

The Maxim devices are pretty straight-forward to convert over to use GPIO descriptors, so let's do it. Tested with some x86_64 allmodconfig and aarch64 allmodconfig to smoke out the worst bugs this time. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Changes in v2: - Rebased on v6.6-rc1 - Link to v1: https://lore.kernel.org/r/20230825-descriptors-asoc-max-v1-0-b212292b2f08@l… --- Linus Walleij (7): ASoC: max9768: Convert to use GPIO descriptors ASoC: max98357a: Drop pointless include ASoC: max98373: Convert to use GPIO descriptors ASoC: max98388: Correct the includes ASoC: max98396: Drop pointless include ASoC: max98520: Drop pointless includes ASoC: max98927: Drop pointless includes include/sound/max9768.h | 4 ---- sound/soc/codecs/max9768.c | 45 +++++++++++++++++++++-------------------- sound/soc/codecs/max98357a.c | 1 - sound/soc/codecs/max98373-i2c.c | 17 ---------------- sound/soc/codecs/max98373.c | 35 +++++++++++++++++--------------- sound/soc/codecs/max98373.h | 2 +- sound/soc/codecs/max98388.c | 3 +-- sound/soc/codecs/max98396.c | 1 - sound/soc/codecs/max98520.c | 2 -- sound/soc/codecs/max98927.c | 2 -- 10 files changed, 44 insertions(+), 68 deletions(-) --- base-commit: 0bb80ecc33a8fb5a682236443c1e740d5c917d1d change-id: 20230817-descriptors-asoc-max-b71ac02bf1ff Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

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[PATCH v3 1/4] ASoC: cs35l45: Checks index of cs35l45_irqs[]
by Vlad Karpovich 12 Sep '23

12 Sep '23

From: Ricardo Rivera-Matos <rriveram(a)opensource.cirrus.com> Checks the index computed by the virq offset before printing the error condition in cs35l45_spk_safe_err() handler. Signed-off-by: Ricardo Rivera-Matos <rriveram(a)opensource.cirrus.com> Signed-off-by: Vlad Karpovich <vkarpovi(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 2ac4612402eb..02b1172d2647 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -1023,7 +1023,10 @@ static irqreturn_t cs35l45_spk_safe_err(int irq, void *data) i = irq - regmap_irq_get_virq(cs35l45->irq_data, 0); - dev_err(cs35l45->dev, "%s condition detected!\n", cs35l45_irqs[i].name); + if (i < 0 || i >= ARRAY_SIZE(cs35l45_irqs)) + dev_err(cs35l45->dev, "Unspecified global error condition (%d) detected!\n", irq); + else + dev_err(cs35l45->dev, "%s condition detected!\n", cs35l45_irqs[i].name); return IRQ_HANDLED; } -- 2.25.1

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[PATCH] ASoC: meson: axg: extend TDM maximum sample rate to 384kHz
by Jerome Brunet 12 Sep '23

12 Sep '23

The TDM HW on the axg SoC families and derivatives actually supports 384kHz sampling rate. Update the fifo and tdm interface constraints accordingly. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/axg-fifo.h | 2 +- sound/soc/meson/axg-tdm.h | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index bccfb770b339..2e3d0108179b 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -31,7 +31,7 @@ static struct snd_pcm_hardware axg_fifo_hw = { SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), .formats = AXG_FIFO_FORMATS, .rate_min = 5512, - .rate_max = 192000, + .rate_max = 384000, .channels_min = 1, .channels_max = AXG_FIFO_CH_MAX, .period_bytes_min = AXG_FIFO_BURST, diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h index b63acd723c87..df528e8cb7c9 100644 --- a/sound/soc/meson/axg-fifo.h +++ b/sound/soc/meson/axg-fifo.h @@ -22,7 +22,7 @@ struct snd_soc_pcm_runtime; #define AXG_FIFO_CH_MAX 128 #define AXG_FIFO_RATES (SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_192000) + SNDRV_PCM_RATE_8000_384000) #define AXG_FIFO_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_LE | \ diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h index 5774ce0916d4..42f7470b9a7f 100644 --- a/sound/soc/meson/axg-tdm.h +++ b/sound/soc/meson/axg-tdm.h @@ -16,7 +16,7 @@ #define AXG_TDM_NUM_LANES 4 #define AXG_TDM_CHANNEL_MAX 128 #define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_192000) + SNDRV_PCM_RATE_8000_384000) #define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_LE | \ -- 2.40.1

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