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

[PATCH AUTOSEL 6.5 08/31] ASoC: core: Do not call link_exit() on uninitialized rtd objects
by Sasha Levin 18 Oct '23

18 Oct '23

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit dd9f9cc1e6b9391140afa5cf27bb47c9e2a08d02 ] On init we have sequence: for_each_card_prelinks(card, i, dai_link) { ret = snd_soc_add_pcm_runtime(card, dai_link); ret = init_some_other_things(...); if (ret) goto probe_end: for_each_card_rtds(card, rtd) { ret = soc_init_pcm_runtime(card, rtd); probe_end: while on exit: for_each_card_rtds(card, rtd) snd_soc_link_exit(rtd); If init_some_other_things() step fails due to error we end up with not fully setup rtds and try to call snd_soc_link_exit on them, which depending on contents on .link_exit handler, can end up dereferencing NULL pointer. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230929103243.705433-2-amadeuszx.slawinski@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 20 +++++++++++++++----- 2 files changed, 17 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index b27f84580c5b0..cf34810882347 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1125,6 +1125,8 @@ struct snd_soc_pcm_runtime { unsigned int pop_wait:1; unsigned int fe_compr:1; /* for Dynamic PCM */ + bool initialized; + int num_components; struct snd_soc_component *components[]; /* CPU/Codec/Platform */ }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1a0bde23f5e6f..2d85164457f73 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1259,7 +1259,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, snd_soc_runtime_get_dai_fmt(rtd); ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt); if (ret) - return ret; + goto err; /* add DPCM sysfs entries */ soc_dpcm_debugfs_add(rtd); @@ -1284,17 +1284,26 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, /* create compress_device if possible */ ret = snd_soc_dai_compress_new(cpu_dai, rtd, num); if (ret != -ENOTSUPP) - return ret; + goto err; /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); - return ret; + goto err; } - return snd_soc_pcm_dai_new(rtd); + ret = snd_soc_pcm_dai_new(rtd); + if (ret < 0) + goto err; + + rtd->initialized = true; + + return 0; +err: + snd_soc_link_exit(rtd); + return ret; } static void soc_set_name_prefix(struct snd_soc_card *card, @@ -1892,7 +1901,8 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) /* release machine specific resources */ for_each_card_rtds(card, rtd) - snd_soc_link_exit(rtd); + if (rtd->initialized) + snd_soc_link_exit(rtd); /* remove and free each DAI */ soc_remove_link_dais(card); soc_remove_link_components(card); -- 2.40.1

1 0

[PATCH AUTOSEL 6.5 07/31] ASoC: fsl-asoc-card: use integer type for fll_id and pll_id
by Sasha Levin 18 Oct '23

18 Oct '23

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 2b21207afd06714986a3d22442ed4860ba4f9ced ] As the pll_id and pll_id can be zero (WM8960_SYSCLK_AUTO) with the commit 2bbc2df46e67 ("ASoC: wm8960: Make automatic the default clocking mode") Then the machine driver will skip to call set_sysclk() and set_pll() for codec, when the sysclk rate is different with what wm8960 read at probe, the output sound frequency is wrong. So change the fll_id and pll_id initial value, still keep machine driver's behavior same as before. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Link: https://lore.kernel.org/r/1695202992-24864-1-git-send-email-shengjiu.wang@n… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 76b5bfc288fde..bab7d34cf585b 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -52,8 +52,8 @@ struct codec_priv { unsigned long mclk_freq; unsigned long free_freq; u32 mclk_id; - u32 fll_id; - u32 pll_id; + int fll_id; + int pll_id; }; /** @@ -206,7 +206,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, } /* Specific configuration for PLL */ - if (codec_priv->pll_id && codec_priv->fll_id) { + if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) pll_out = priv->sample_rate * 384; else @@ -248,7 +248,7 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) priv->streams &= ~BIT(substream->stream); - if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) { + if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { /* Force freq to be free_freq to avoid error message in codec */ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), codec_priv->mclk_id, @@ -621,6 +621,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dapm_routes = audio_map; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.driver_name = DRIVER_NAME; + + priv->codec_priv.fll_id = -1; + priv->codec_priv.pll_id = -1; + /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; -- 2.40.1

1 0

[PATCH AUTOSEL 6.5 04/31] ASoC: simple-card: fixup asoc_simple_probe() error handling
by Sasha Levin 18 Oct '23

18 Oct '23

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 41bae58df411f9accf01ea660730649b2fab1dab ] asoc_simple_probe() is used for both "DT probe" (A) and "platform probe" (B). It uses "goto err" when error case, but it is not needed for "platform probe" case (B). Thus it is using "return" directly there. static int asoc_simple_probe(...) { ^ if (...) { | ... (A) if (ret < 0) | goto err; v } else { ^ ... | if (ret < 0) (B) return -Exxx; v } ... ^ if (ret < 0) (C) goto err; v ... err: (D) simple_util_clean_reference(card); return ret; } Both case are using (C) part, and it calls (D) when err case. But (D) will do nothing for (B) case. Because of these behavior, current code itself is not wrong, but is confusable, and more, static analyzing tool will warning on (B) part (should use goto err). To avoid static analyzing tool warning, this patch uses "goto err" on (B) part. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87o7hy7mlh.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/simple-card.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0745bf6a09aa0..5005d3c9c659a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -701,10 +701,12 @@ static int asoc_simple_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link = priv->dai_link; struct simple_dai_props *dai_props = priv->dai_props; + ret = -EINVAL; + cinfo = dev->platform_data; if (!cinfo) { dev_err(dev, "no info for asoc-simple-card\n"); - return -EINVAL; + goto err; } if (!cinfo->name || @@ -713,7 +715,7 @@ static int asoc_simple_probe(struct platform_device *pdev) !cinfo->platform || !cinfo->cpu_dai.name) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); - return -EINVAL; + goto err; } cpus = dai_link->cpus; -- 2.40.1

1 0

[PATCH AUTOSEL 6.5 03/31] ASoC: Intel: sof_sdw: add support for SKU 0B14
by Sasha Levin 18 Oct '23

18 Oct '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit fb0b8d299781be8d46b3612aa96cef28da0d93f4 ] One more missing SKU in the list. Closes: https://github.com/thesofproject/linux/issues/4543 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230919092125.1922468-1-yung-chuan.liao@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index c86f8f9a61003..7d9b3a07945d7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -366,6 +366,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B14"), + }, + /* No Jack */ + .driver_data = (void *)SOF_SDW_TGL_HDMI, + }, + { .callback = sof_sdw_quirk_cb, .matches = { -- 2.40.1

1 0

[PATCH AUTOSEL 6.5 02/31] ASoC: Intel: soc-acpi: Add entry for sof_es8336 in MTL match table.
by Sasha Levin 18 Oct '23

18 Oct '23

From: Balamurugan C <balamurugan.c(a)intel.com> [ Upstream commit 381ddcd5875e496f2eae06bb65853271b7150fee ] Adding support for ES83x6 codec in MTL match table. Signed-off-by: Balamurugan C <balamurugan.c(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230919091136.1922253-2-yung-chuan.liao@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index ed9821adc1d9d..18da4b7e378d1 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -30,6 +30,11 @@ static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = { .codecs = {"10EC5682", "RTL5682"}, }; +static const struct snd_soc_acpi_codecs mtl_essx_83x6 = { + .num_codecs = 3, + .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { { .comp_ids = &mtl_rt5682_rt5682s_hp, @@ -52,6 +57,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .quirk_data = &mtl_rt1019p_amp, .sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg", }, + { + .comp_ids = &mtl_essx_83x6, + .drv_name = "sof-essx8336", + .sof_tplg_filename = "sof-mtl-es8336", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | + SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | + SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); -- 2.40.1

1 0

USB-Audio: Add FiiO K11 config
by GitHub pull_request - edited 18 Oct '23

18 Oct '23

alsa-project/alsa-ucm-conf pull request #360 was edited from ZVNexus: Without an ALSA config for the DAC autosuspend seems to be broken, every 10 seconds after pausing audio playback there is a 1 second delay to start it back up. Added a config for it and now there is no delay. Let me know if something else would make more sense. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/360 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/360.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RFC PATCH] ALSA: virtio: use copy and fill_silence callbacks
by Matias Ezequiel Vara Larsen 18 Oct '23

18 Oct '23

This commit replaces the mmap mechanism with the copy() and fill_silence() callbacks for both capturing and playback for the virtio-sound driver. This change is required to prevent the updating of the content of a buffer that is already in the available ring. The current mechanism splits a dma buffer into descriptors that are exposed to the device. This dma buffer is shared with the user application. When the device consumes a buffer, the driver moves the request from the used ring to available ring. The driver exposes the buffer to the device without knowing if the content has been updated from the user. The section 2.8.21.1 of the virtio spec states that: "The device MAY access the descriptor chains the driver created and the memory they refer to immediately". If the device picks up buffers from the available ring just after it is notified, it happens that the content may be old. By providing the copy() callback, the driver first updates the content of the buffer, and then, exposes the buffer to the device by enqueuing it in the available ring. Thus, device always picks up a buffer that is updated. For capturing, the driver starts by exposing all the available buffers to device. After device updates the content of a buffer, it enqueues it in the used ring. It is only after the copy() for capturing is issued that the driver re-enqueues the buffer in the available ring. Note that the copy() function assumes that user is always writing a period. Testing shows that this is true but I may be wrong. This RFC aims at clarifying this. Signed-off-by: Matias Ezequiel Vara Larsen <mvaralar(a)redhat.com> --- sound/virtio/virtio_pcm.c | 11 ++-- sound/virtio/virtio_pcm.h | 9 +++- sound/virtio/virtio_pcm_msg.c | 50 ++++++++++++++++--- sound/virtio/virtio_pcm_ops.c | 94 +++++++++++++++++++++++++++++++---- 4 files changed, 137 insertions(+), 27 deletions(-) diff --git a/sound/virtio/virtio_pcm.c b/sound/virtio/virtio_pcm.c index c10d91fff2fb..bfe982952303 100644 --- a/sound/virtio/virtio_pcm.c +++ b/sound/virtio/virtio_pcm.c @@ -104,8 +104,6 @@ static int virtsnd_pcm_build_hw(struct virtio_pcm_substream *vss, * only message-based transport. */ vss->hw.info = - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_INTERLEAVED | @@ -471,12 +469,11 @@ int virtsnd_pcm_build_devs(struct virtio_snd *snd) for (kss = ks->substream; kss; kss = kss->next) vs->substreams[kss->number]->substream = kss; - snd_pcm_set_ops(vpcm->pcm, i, &virtsnd_pcm_ops); + if (i == SNDRV_PCM_STREAM_CAPTURE) + snd_pcm_set_ops(vpcm->pcm, i, &virtsnd_pcm_capture_ops); + else + snd_pcm_set_ops(vpcm->pcm, i, &virtsnd_pcm_playback_ops); } - - snd_pcm_set_managed_buffer_all(vpcm->pcm, - SNDRV_DMA_TYPE_VMALLOC, NULL, - 0, 0); } return 0; diff --git a/sound/virtio/virtio_pcm.h b/sound/virtio/virtio_pcm.h index 062eb8e8f2cf..1c1106ec971f 100644 --- a/sound/virtio/virtio_pcm.h +++ b/sound/virtio/virtio_pcm.h @@ -50,6 +50,8 @@ struct virtio_pcm_substream { struct work_struct elapsed_period; spinlock_t lock; size_t buffer_bytes; + u8 *buffer; + size_t buffer_sz; size_t hw_ptr; bool xfer_enabled; bool xfer_xrun; @@ -90,7 +92,8 @@ struct virtio_pcm { struct virtio_pcm_stream streams[SNDRV_PCM_STREAM_LAST + 1]; }; -extern const struct snd_pcm_ops virtsnd_pcm_ops; +extern const struct snd_pcm_ops virtsnd_pcm_playback_ops; +extern const struct snd_pcm_ops virtsnd_pcm_capture_ops; int virtsnd_pcm_validate(struct virtio_device *vdev); @@ -117,7 +120,9 @@ int virtsnd_pcm_msg_alloc(struct virtio_pcm_substream *vss, void virtsnd_pcm_msg_free(struct virtio_pcm_substream *vss); -int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss); +int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss, bool single); + +int virtsnd_pcm_msg_send_locked(struct virtio_pcm_substream *vss, bool single); unsigned int virtsnd_pcm_msg_pending_num(struct virtio_pcm_substream *vss); diff --git a/sound/virtio/virtio_pcm_msg.c b/sound/virtio/virtio_pcm_msg.c index aca2dc1989ba..9a5f9814cb62 100644 --- a/sound/virtio/virtio_pcm_msg.c +++ b/sound/virtio/virtio_pcm_msg.c @@ -132,7 +132,6 @@ static void virtsnd_pcm_sg_from(struct scatterlist *sgs, int nsgs, u8 *data, int virtsnd_pcm_msg_alloc(struct virtio_pcm_substream *vss, unsigned int periods, unsigned int period_bytes) { - struct snd_pcm_runtime *runtime = vss->substream->runtime; unsigned int i; vss->msgs = kcalloc(periods, sizeof(*vss->msgs), GFP_KERNEL); @@ -142,7 +141,7 @@ int virtsnd_pcm_msg_alloc(struct virtio_pcm_substream *vss, vss->nmsgs = periods; for (i = 0; i < periods; ++i) { - u8 *data = runtime->dma_area + period_bytes * i; + u8 *data = vss->buffer + period_bytes * i; int sg_num = virtsnd_pcm_sg_num(data, period_bytes); struct virtio_pcm_msg *msg; @@ -186,10 +185,12 @@ void virtsnd_pcm_msg_free(struct virtio_pcm_substream *vss) /** * virtsnd_pcm_msg_send() - Send asynchronous I/O messages. * @vss: VirtIO PCM substream. + * @single: true to enqueue a single message, false to enqueue all of them. * * All messages are organized in an ordered circular list. Each time the - * function is called, all currently non-enqueued messages are added to the - * virtqueue. For this, the function keeps track of two values: + * function is called, first non-enqueued message is added to the virtqueue. + * When single is True, only the first message is enqueued. When False, all the + * available messages are enqueued. The function keeps track of two values: * * msg_last_enqueued = index of the last enqueued message, * msg_count = # of pending messages in the virtqueue. @@ -198,7 +199,7 @@ void virtsnd_pcm_msg_free(struct virtio_pcm_substream *vss) * spinlocks to be held by caller. * Return: 0 on success, -errno on failure. */ -int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss) +int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss, bool single) { struct snd_pcm_runtime *runtime = vss->substream->runtime; struct virtio_snd *snd = vss->snd; @@ -211,6 +212,13 @@ int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss) i = (vss->msg_last_enqueued + 1) % runtime->periods; n = runtime->periods - vss->msg_count; + if (single) { + if (n < 1) + return -EFAULT; + + n = 1; + } + for (; n; --n, i = (i + 1) % runtime->periods) { struct virtio_pcm_msg *msg = vss->msgs[i]; struct scatterlist *psgs[] = { @@ -250,6 +258,36 @@ int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss) return 0; } +/** + * virtsnd_pcm_msg_send_locked() - Send asynchronous I/O messages. + * @vss: VirtIO PCM substream. + * @single: true to enqueue a single message, false to enqueue all of them. + * + * This function holds the tx/rx queue and the VirtIO substream spinlocks + * before calling virtsnd_pcm_msg_send(). This is a wrapper function to ease + * the invocation of virtsnd_pcm_msg_send(). + * + * Context: Any context. + * Return: 0 on success, -errno on failure. + */ + +int virtsnd_pcm_msg_send_locked(struct virtio_pcm_substream *vss, bool single) +{ + struct virtio_snd_queue *queue; + int rc; + unsigned long flags; + + queue = virtsnd_pcm_queue(vss); + + spin_lock_irqsave(&queue->lock, flags); + spin_lock(&vss->lock); + rc = virtsnd_pcm_msg_send(vss, single); + spin_unlock(&vss->lock); + spin_unlock_irqrestore(&queue->lock, flags); + + return rc; +} + /** * virtsnd_pcm_msg_pending_num() - Returns the number of pending I/O messages. * @vss: VirtIO substream. @@ -320,8 +358,6 @@ static void virtsnd_pcm_msg_complete(struct virtio_pcm_msg *msg, le32_to_cpu(msg->status.latency_bytes)); schedule_work(&vss->elapsed_period); - - virtsnd_pcm_msg_send(vss); } else if (!vss->msg_count) { wake_up_all(&vss->msg_empty); } diff --git a/sound/virtio/virtio_pcm_ops.c b/sound/virtio/virtio_pcm_ops.c index f8bfb87624be..a208439dbff8 100644 --- a/sound/virtio/virtio_pcm_ops.c +++ b/sound/virtio/virtio_pcm_ops.c @@ -238,6 +238,11 @@ static int virtsnd_pcm_hw_params(struct snd_pcm_substream *substream, */ virtsnd_pcm_msg_free(vss); + vss->buffer_sz = params_buffer_bytes(hw_params); + vss->buffer = alloc_pages_exact(vss->buffer_sz, GFP_KERNEL); + if (!vss->buffer) + return -ENOMEM; + return virtsnd_pcm_msg_alloc(vss, params_periods(hw_params), params_period_bytes(hw_params)); } @@ -257,6 +262,11 @@ static int virtsnd_pcm_hw_free(struct snd_pcm_substream *substream) if (!virtsnd_pcm_msg_pending_num(vss)) virtsnd_pcm_msg_free(vss); + if (vss->buffer) { + free_pages_exact(vss->buffer, vss->buffer_sz); + vss->buffer = NULL; + } + return 0; } @@ -331,15 +341,18 @@ static int virtsnd_pcm_trigger(struct snd_pcm_substream *substream, int command) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: queue = virtsnd_pcm_queue(vss); - spin_lock_irqsave(&queue->lock, flags); - spin_lock(&vss->lock); - rc = virtsnd_pcm_msg_send(vss); - if (!rc) - vss->xfer_enabled = true; - spin_unlock(&vss->lock); - spin_unlock_irqrestore(&queue->lock, flags); - if (rc) - return rc; + // The buffers should be exposed first during capturing so that + // the device can consume them. Capturing cannot begin + // otherwise. + if (vss->direction == SNDRV_PCM_STREAM_CAPTURE) { + rc = virtsnd_pcm_msg_send_locked(vss, false); + if (rc) + return rc; + } + + spin_lock_irqsave(&vss->lock, flags); + vss->xfer_enabled = true; + spin_unlock_irqrestore(&vss->lock, flags); msg = virtsnd_pcm_ctl_msg_alloc(vss, VIRTIO_SND_R_PCM_START, GFP_KERNEL); @@ -450,8 +463,66 @@ virtsnd_pcm_pointer(struct snd_pcm_substream *substream) return hw_ptr; } -/* PCM substream operators map. */ -const struct snd_pcm_ops virtsnd_pcm_ops = { +static int virtsnd_pcm_pb_copy(struct snd_pcm_substream *substream, + int channel, unsigned long pos, struct iov_iter + *src, unsigned long count) +{ + struct virtio_pcm_substream *vss = snd_pcm_substream_chip(substream); + + if (unlikely(pos + count > vss->buffer_sz)) + return -EINVAL; + + if (copy_from_iter(vss->buffer + pos, count, src) != count) + return -EFAULT; + + return virtsnd_pcm_msg_send_locked(vss, true); +} + +static int virtsnd_pcm_cap_copy(struct snd_pcm_substream *substream, + int channel, unsigned long pos, struct iov_iter + *dst, unsigned long count) +{ + struct virtio_pcm_substream *vss = snd_pcm_substream_chip(substream); + + if (unlikely(pos + count > vss->buffer_sz)) + return -EINVAL; + + if (copy_to_iter(vss->buffer + pos, count, dst) != count) + return -EFAULT; + + return virtsnd_pcm_msg_send_locked(vss, true); +} + +static int virtsnd_pcm_pb_silence(struct snd_pcm_substream *substream, int channel, + unsigned long pos, unsigned long count) +{ + struct virtio_pcm_substream *vss = snd_pcm_substream_chip(substream); + + if (unlikely(pos + count > vss->buffer_sz)) + return -EINVAL; + + memset(vss->buffer + pos, 0, count); + + return virtsnd_pcm_msg_send_locked(vss, true); +} + +/* PCM substream operators map for playback. */ +const struct snd_pcm_ops virtsnd_pcm_playback_ops = { + .open = virtsnd_pcm_open, + .close = virtsnd_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = virtsnd_pcm_hw_params, + .hw_free = virtsnd_pcm_hw_free, + .prepare = virtsnd_pcm_prepare, + .trigger = virtsnd_pcm_trigger, + .sync_stop = virtsnd_pcm_sync_stop, + .pointer = virtsnd_pcm_pointer, + .copy = virtsnd_pcm_pb_copy, + .fill_silence = virtsnd_pcm_pb_silence, +}; + +/* PCM substream operators map for capturing. */ +const struct snd_pcm_ops virtsnd_pcm_capture_ops = { .open = virtsnd_pcm_open, .close = virtsnd_pcm_close, .ioctl = snd_pcm_lib_ioctl, @@ -461,4 +532,5 @@ const struct snd_pcm_ops virtsnd_pcm_ops = { .trigger = virtsnd_pcm_trigger, .sync_stop = virtsnd_pcm_sync_stop, .pointer = virtsnd_pcm_pointer, + .copy = virtsnd_pcm_cap_copy, }; base-commit: 8a749fd1a8720d4619c91c8b6e7528c0a355c0aa -- 2.41.0

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[PATCH][next] ASoC: tas2781: make const read-only array magic_number static
by Colin Ian King 18 Oct '23

18 Oct '23

Don't populate the const read-only array magic_number on the stack, instead make it static const. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/tas2781-fmwlib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index e27775d834e9..4efe95b60aaa 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1757,7 +1757,7 @@ static int fw_parse_header(struct tasdevice_priv *tas_priv, { struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); - const unsigned char magic_number[] = { 0x35, 0x35, 0x35, 0x32 }; + static const unsigned char magic_number[] = { 0x35, 0x35, 0x35, 0x32 }; const unsigned char *buf = (unsigned char *)fmw->data; if (offset + 92 > fmw->size) { -- 2.39.2

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[PATCH] ASoC: amd: ps: enable wake capability for acp pci driver
by Vijendar Mukunda 17 Oct '23

17 Oct '23

Enable wake capability for acp pci driver for Pink Sardine platform. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 7e4c0ec9e56c..5927eef04170 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -650,6 +650,7 @@ static int snd_acp63_probe(struct pci_dev *pci, goto de_init; } skip_pdev_creation: + device_set_wakeup_enable(&pci->dev, true); pm_runtime_set_autosuspend_delay(&pci->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pci->dev); pm_runtime_put_noidle(&pci->dev); -- 2.34.1

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[PATCH] ASoC: dt-bindings: tas5805m: Disallow undefined properties
by Rob Herring 17 Oct '23

17 Oct '23

Device specific bindings should not allow undefined properties. This is accomplished in json-schema with 'additionalProperties: false'. Examples should be last in the schema, so move additionalProperties up while we're here. Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/tas5805m.yaml | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas5805m.yaml b/Documentation/devicetree/bindings/sound/tas5805m.yaml index 63edf52f061c..12c41974274e 100644 --- a/Documentation/devicetree/bindings/sound/tas5805m.yaml +++ b/Documentation/devicetree/bindings/sound/tas5805m.yaml @@ -37,6 +37,8 @@ properties: generated from TI's PPC3 tool. $ref: /schemas/types.yaml#/definitions/string +additionalProperties: false + examples: - | i2c { @@ -52,5 +54,4 @@ examples: ti,dsp-config-name = "mono_pbtl_48khz"; }; }; - -additionalProperties: true +... -- 2.42.0

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