From 63f5235e0291152a2ac2c4ef3c1196cb6dfb3ef7 Mon Sep 17 00:00:00 2001 From: Chris Chiu Date: Wed, 30 Apr 2025 18:18:43 +0800 Subject: [PATCH 01/16] ALSA: hda/realtek - Add more HP laptops which need mute led fixup More HP EliteBook with Realtek HDA codec ALC3247 and combined CS35L56 Amplifiers need quirk ALC236_FIXUP_HP_GPIO_LED to fix the micmute LED. Signed-off-by: Chris Chiu Cc: Link: https://patch.msgid.link/20250430101843.150833-1-chris.chiu@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7810d5f9b5aa..8a2b09e4a7d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10863,8 +10863,11 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8de8, "HP Gemtree", ALC245_FIXUP_TAS2781_SPI_2), SND_PCI_QUIRK(0x103c, 0x8de9, "HP Gemtree", ALC245_FIXUP_TAS2781_SPI_2), SND_PCI_QUIRK(0x103c, 0x8dec, "HP EliteBook 640 G12", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ded, "HP EliteBook 640 G12", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8dee, "HP EliteBook 660 G12", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8def, "HP EliteBook 660 G12", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8df0, "HP EliteBook 630 G12", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8df1, "HP EliteBook 630 G12", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8dfc, "HP EliteBook 645 G12", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8dfe, "HP EliteBook 665 G12", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8e11, "HP Trekker", ALC287_FIXUP_CS35L41_I2C_2), -- 2.51.0 From edea92770a3b6454dc796fc5436a3315bb402181 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 30 Apr 2025 18:52:08 +0200 Subject: [PATCH 02/16] ASoC: stm32: sai: skip useless iterations on kernel rate loop the frequency of the kernel clock must be greater than or equal to the bitclock rate. When searching for a convenient kernel clock rate in stm32_sai_set_parent_rate() function, it is useless to continue the loop below bitclock rate, as it will result in a invalid kernel clock rate. Change the loop output condition. Fixes: 2cfe1ff22555 ("ASoC: stm32: sai: add stm32mp25 support") Signed-off-by: Olivier Moysan Link: https://patch.msgid.link/20250430165210.321273-2-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index e8c1abf1ae0a..4d018b4bc3f0 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -409,11 +409,11 @@ static int stm32_sai_set_parent_rate(struct stm32_sai_sub_data *sai, unsigned int rate) { struct platform_device *pdev = sai->pdev; - unsigned int sai_ck_rate, sai_ck_max_rate, sai_curr_rate, sai_new_rate; + unsigned int sai_ck_rate, sai_ck_max_rate, sai_ck_min_rate, sai_curr_rate, sai_new_rate; int div, ret; /* - * Set maximum expected kernel clock frequency + * Set minimum and maximum expected kernel clock frequency * - mclk on or spdif: * f_sai_ck = MCKDIV * mclk-fs * fs * Here typical 256 ratio is assumed for mclk-fs @@ -423,13 +423,16 @@ static int stm32_sai_set_parent_rate(struct stm32_sai_sub_data *sai, * Set constraint MCKDIV * FRL <= 256, to ensure MCKDIV is in available range * f_sai_ck = sai_ck_max_rate * pow_of_two(FRL) / 256 */ + sai_ck_min_rate = rate * 256; if (!(rate % SAI_RATE_11K)) sai_ck_max_rate = SAI_MAX_SAMPLE_RATE_11K * 256; else sai_ck_max_rate = SAI_MAX_SAMPLE_RATE_8K * 256; - if (!sai->sai_mclk && !STM_SAI_PROTOCOL_IS_SPDIF(sai)) + if (!sai->sai_mclk && !STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + sai_ck_min_rate = rate * sai->fs_length; sai_ck_max_rate /= DIV_ROUND_CLOSEST(256, roundup_pow_of_two(sai->fs_length)); + } /* * Request exclusivity, as the clock is shared by SAI sub-blocks and by @@ -472,7 +475,7 @@ static int stm32_sai_set_parent_rate(struct stm32_sai_sub_data *sai, /* Try a lower frequency */ div++; sai_ck_rate = sai_ck_max_rate / div; - } while (sai_ck_rate > rate); + } while (sai_ck_rate >= sai_ck_min_rate); /* No accurate rate found */ dev_err(&pdev->dev, "Failed to find an accurate rate"); -- 2.51.0 From cce34d113e2a592806abcdc02c7f8513775d8b20 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 30 Apr 2025 18:52:09 +0200 Subject: [PATCH 03/16] ASoC: stm32: sai: add a check on minimal kernel frequency On MP2 SoCs SAI kernel clock rate is managed through stm32_sai_set_parent_rate() function. If the kernel clock rate was set previously to a low frequency, this frequency may be too low to support the newly requested audio stream rate. However the stm32_sai_rate_accurate() will only check accuracy against the maximum kernel clock rate. The function will return leaving the kernel clock rate unchanged. Add a check on minimal frequency requirement, to avoid this. Fixes: 2cfe1ff22555 ("ASoC: stm32: sai: add stm32mp25 support") Signed-off-by: Olivier Moysan Link: https://patch.msgid.link/20250430165210.321273-3-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 4d018b4bc3f0..bf5299ba11c3 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -447,7 +447,10 @@ static int stm32_sai_set_parent_rate(struct stm32_sai_sub_data *sai, * return immediately. */ sai_curr_rate = clk_get_rate(sai->sai_ck); - if (stm32_sai_rate_accurate(sai_ck_max_rate, sai_curr_rate)) + dev_dbg(&pdev->dev, "kernel clock rate: min [%u], max [%u], current [%u]", + sai_ck_min_rate, sai_ck_max_rate, sai_curr_rate); + if (stm32_sai_rate_accurate(sai_ck_max_rate, sai_curr_rate) && + sai_curr_rate >= sai_ck_min_rate) return 0; /* -- 2.51.0 From 02b44a2b2bdcee03cbb92484d31e9ca1b91b2a38 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Wed, 30 Apr 2025 11:31:19 +0100 Subject: [PATCH 04/16] ASoC: intel/sdw_utils: Add volume limit to cs42l43 speakers The volume control for cs42l43 speakers has a maximum gain of +31.5 dB. However, for many use cases, this can cause distorted audio, depending various factors, such as other signal-processing elements in the chain, for example if the audio passes through a gain control before reaching the codec or the signal path has been tuned for a particular maximum gain in the codec. In the case of systems which use the soc_sdw_cs42l43 driver, audio will likely be distorted in all cases above 0 dB, therefore add a volume limit of 128, which is 0 dB maximum volume inside this driver. Signed-off-by: Stefan Binding Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20250430103134.24579-2-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sdw_utils/soc_sdw_cs42l43.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/sdw_utils/soc_sdw_cs42l43.c b/sound/soc/sdw_utils/soc_sdw_cs42l43.c index 668c9d28a1c1..b415d45d520d 100644 --- a/sound/soc/sdw_utils/soc_sdw_cs42l43.c +++ b/sound/soc/sdw_utils/soc_sdw_cs42l43.c @@ -20,6 +20,8 @@ #include #include +#define CS42L43_SPK_VOLUME_0DB 128 /* 0dB Max */ + static const struct snd_soc_dapm_route cs42l43_hs_map[] = { { "Headphone", NULL, "cs42l43 AMP3_OUT" }, { "Headphone", NULL, "cs42l43 AMP4_OUT" }, @@ -117,6 +119,14 @@ int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_so return -ENOMEM; } + ret = snd_soc_limit_volume(card, "cs42l43 Speaker Digital Volume", + CS42L43_SPK_VOLUME_0DB); + if (ret) + dev_err(card->dev, "cs42l43 speaker volume limit failed: %d\n", ret); + else + dev_info(card->dev, "Setting CS42L43 Speaker volume limit to %d\n", + CS42L43_SPK_VOLUME_0DB); + ret = snd_soc_dapm_add_routes(&card->dapm, cs42l43_spk_map, ARRAY_SIZE(cs42l43_spk_map)); if (ret) -- 2.51.0 From d5463e531c128ff1b141fdba2e13345cd50028a4 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Wed, 30 Apr 2025 11:31:20 +0100 Subject: [PATCH 05/16] ASoC: intel/sdw_utils: Add volume limit to cs35l56 speakers The volume control for cs35l56 speakers has a maximum gain of +12 dB. However, for many use cases, this can cause distorted audio, depending various factors, such as other signal-processing elements in the chain, for example if the audio passes through a gain control before reaching the amp or the signal path has been tuned for a particular maximum gain in the amp. In the case of systems which use the soc_sdw_* driver, audio will likely be distorted in all cases above 0 dB, therefore add a volume limit of 400, which is 0 dB maximum volume inside this driver. The volume limit should be applied to both soundwire and soundwire bridge configurations. Signed-off-by: Stefan Binding Link: https://patch.msgid.link/20250430103134.24579-3-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 1 + sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c | 4 ++++ sound/soc/sdw_utils/soc_sdw_cs_amp.c | 24 ++++++++++++++++++++ 3 files changed, 29 insertions(+) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 36a4a1e1d8ca..d8bd5d37131a 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -226,6 +226,7 @@ int asoc_sdw_cs_amp_init(struct snd_soc_card *card, bool playback); int asoc_sdw_cs_spk_feedback_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs35l56_volume_limit(struct snd_soc_card *card, const char *name_prefix); /* MAXIM codec support */ int asoc_sdw_maxim_init(struct snd_soc_card *card, diff --git a/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c b/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c index 246e5c2e0af5..c7e55f443351 100644 --- a/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c +++ b/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c @@ -60,6 +60,10 @@ static int asoc_sdw_bridge_cs35l56_asp_init(struct snd_soc_pcm_runtime *rtd) /* 4 x 16-bit sample slots and FSYNC=48000, BCLK=3.072 MHz */ for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = asoc_sdw_cs35l56_volume_limit(card, codec_dai->component->name_prefix); + if (ret) + return ret; + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_mask, rx_mask, 4, 16); if (ret < 0) return ret; diff --git a/sound/soc/sdw_utils/soc_sdw_cs_amp.c b/sound/soc/sdw_utils/soc_sdw_cs_amp.c index 4b6181cf2971..35b550bcd4de 100644 --- a/sound/soc/sdw_utils/soc_sdw_cs_amp.c +++ b/sound/soc/sdw_utils/soc_sdw_cs_amp.c @@ -16,6 +16,25 @@ #define CODEC_NAME_SIZE 8 #define CS_AMP_CHANNELS_PER_AMP 4 +#define CS35L56_SPK_VOLUME_0DB 400 /* 0dB Max */ + +int asoc_sdw_cs35l56_volume_limit(struct snd_soc_card *card, const char *name_prefix) +{ + char *volume_ctl_name; + int ret; + + volume_ctl_name = kasprintf(GFP_KERNEL, "%s Speaker Volume", name_prefix); + if (!volume_ctl_name) + return -ENOMEM; + + ret = snd_soc_limit_volume(card, volume_ctl_name, CS35L56_SPK_VOLUME_0DB); + if (ret) + dev_err(card->dev, "%s limit set failed: %d\n", volume_ctl_name, ret); + + kfree(volume_ctl_name); + return ret; +} +EXPORT_SYMBOL_NS(asoc_sdw_cs35l56_volume_limit, "SND_SOC_SDW_UTILS"); int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { @@ -40,6 +59,11 @@ int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai snprintf(widget_name, sizeof(widget_name), "%s SPK", codec_dai->component->name_prefix); + + ret = asoc_sdw_cs35l56_volume_limit(card, codec_dai->component->name_prefix); + if (ret) + return ret; + ret = snd_soc_dapm_add_routes(&card->dapm, &route, 1); if (ret) return ret; -- 2.51.0 From 3cc393d2232ec770b5f79bf0673d67702a3536c3 Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Tue, 29 Apr 2025 11:49:10 +0200 Subject: [PATCH 06/16] ASoC: simple-card-utils: Fix pointer check in graph_util_parse_link_direction Actually check if the passed pointers are valid, before writing to them. This also fixes a USBAN warning: UBSAN: invalid-load in ../sound/soc/fsl/imx-card.c:687:25 load of value 255 is not a valid value for type '_Bool' This is because playback_only is uninitialized and is not written to, as the playback-only property is absent. Fixes: 844de7eebe97 ("ASoC: audio-graph-card2: expand dai_link property part") Signed-off-by: Alexander Stein Link: https://patch.msgid.link/20250429094910.1150970-1-alexander.stein@ew.tq-group.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index a1ccc300e68c..3ae2a212a2e3 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1174,9 +1174,9 @@ void graph_util_parse_link_direction(struct device_node *np, bool is_playback_only = of_property_read_bool(np, "playback-only"); bool is_capture_only = of_property_read_bool(np, "capture-only"); - if (is_playback_only) + if (playback_only) *playback_only = is_playback_only; - if (is_capture_only) + if (capture_only) *capture_only = is_capture_only; } EXPORT_SYMBOL_GPL(graph_util_parse_link_direction); -- 2.51.0 From 7f91f012c1df07af6b915d1f8cece202774bb50e Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 May 2025 01:24:43 +0530 Subject: [PATCH 07/16] ASoC: amd: ps: fix for irq handler return status If any Soundwire manager interrupt is reported, and wake interrupt is not reported, in this scenario irq_flag will be set to zero, which results in interrupt handler return status as IRQ_NONE. Add new irq flag 'wake_irq_flag' check for SoundWire wake interrupt handling to fix incorrect irq handling return status. Fixes: 3898b189079c8 ("ASoC: amd: ps: add soundwire wake interrupt handling") Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20250430195517.3065308-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/pci-ps.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 8e57f31ef7f7..7936b3173632 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -193,6 +193,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) struct amd_sdw_manager *amd_manager; u32 ext_intr_stat, ext_intr_stat1; u16 irq_flag = 0; + u16 wake_irq_flag = 0; u16 sdw_dma_irq_flag = 0; adata = dev_id; @@ -231,7 +232,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) } if (adata->acp_rev >= ACP70_PCI_REV) - irq_flag = check_and_handle_acp70_sdw_wake_irq(adata); + wake_irq_flag = check_and_handle_acp70_sdw_wake_irq(adata); if (ext_intr_stat & BIT(PDM_DMA_STAT)) { ps_pdm_data = dev_get_drvdata(&adata->pdm_dev->dev); @@ -245,7 +246,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) if (sdw_dma_irq_flag) return IRQ_WAKE_THREAD; - if (irq_flag) + if (irq_flag | wake_irq_flag) return IRQ_HANDLED; else return IRQ_NONE; -- 2.51.0 From ff7b190aef6cccdb6f14d20c5753081fe6420e0b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 11 May 2025 15:45:27 +0200 Subject: [PATCH 08/16] ALSA: seq: Fix delivery of UMP events to group ports When an event with UMP message is sent to a UMP client, the EP port receives always no matter where the event is sent to, as it's a catch-all port. OTOH, if an event is sent to EP port, and if the event has a certain UMP Group, it should have been delivered to the associated UMP Group port, too, but this was ignored, so far. This patch addresses the behavior. Now a UMP event sent to the Endpoint port will be delivered to the subscribers of the UMP group port the event is associated with. The patch also does a bit of refactoring to simplify the code about __deliver_to_subscribers(). Fixes: 177ccf811df4 ("ALSA: seq: Support MIDI 2.0 UMP Endpoint port") Link: https://patch.msgid.link/20250511134528.6314-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 52 ++++++++++++++++++++------------ sound/core/seq/seq_ump_convert.c | 18 +++++++++++ sound/core/seq/seq_ump_convert.h | 1 + 3 files changed, 52 insertions(+), 19 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 198c598a5393..880240924bfd 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -732,15 +732,21 @@ static int snd_seq_deliver_single_event(struct snd_seq_client *client, */ static int __deliver_to_subscribers(struct snd_seq_client *client, struct snd_seq_event *event, - struct snd_seq_client_port *src_port, - int atomic, int hop) + int port, int atomic, int hop) { + struct snd_seq_client_port *src_port; struct snd_seq_subscribers *subs; int err, result = 0, num_ev = 0; union __snd_seq_event event_saved; size_t saved_size; struct snd_seq_port_subs_info *grp; + if (port < 0) + return 0; + src_port = snd_seq_port_use_ptr(client, port); + if (!src_port) + return 0; + /* save original event record */ saved_size = snd_seq_event_packet_size(event); memcpy(&event_saved, event, saved_size); @@ -775,6 +781,7 @@ static int __deliver_to_subscribers(struct snd_seq_client *client, read_unlock(&grp->list_lock); else up_read(&grp->list_mutex); + snd_seq_port_unlock(src_port); memcpy(event, &event_saved, saved_size); return (result < 0) ? result : num_ev; } @@ -783,25 +790,32 @@ static int deliver_to_subscribers(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - struct snd_seq_client_port *src_port; - int ret = 0, ret2; - - src_port = snd_seq_port_use_ptr(client, event->source.port); - if (src_port) { - ret = __deliver_to_subscribers(client, event, src_port, atomic, hop); - snd_seq_port_unlock(src_port); - } - - if (client->ump_endpoint_port < 0 || - event->source.port == client->ump_endpoint_port) - return ret; + int ret; +#if IS_ENABLED(CONFIG_SND_SEQ_UMP) + int ret2; +#endif - src_port = snd_seq_port_use_ptr(client, client->ump_endpoint_port); - if (!src_port) + ret = __deliver_to_subscribers(client, event, + event->source.port, atomic, hop); +#if IS_ENABLED(CONFIG_SND_SEQ_UMP) + if (!snd_seq_client_is_ump(client) || client->ump_endpoint_port < 0) return ret; - ret2 = __deliver_to_subscribers(client, event, src_port, atomic, hop); - snd_seq_port_unlock(src_port); - return ret2 < 0 ? ret2 : ret; + /* If it's an event from EP port (and with a UMP group), + * deliver to subscribers of the corresponding UMP group port, too. + * Or, if it's from non-EP port, deliver to subscribers of EP port, too. + */ + if (event->source.port == client->ump_endpoint_port) + ret2 = __deliver_to_subscribers(client, event, + snd_seq_ump_group_port(event), + atomic, hop); + else + ret2 = __deliver_to_subscribers(client, event, + client->ump_endpoint_port, + atomic, hop); + if (ret2 < 0) + return ret2; +#endif + return ret; } /* deliver an event to the destination port(s). diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index ff7e558b4d51..db2f169cae11 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -1285,3 +1285,21 @@ int snd_seq_deliver_to_ump(struct snd_seq_client *source, else return cvt_to_ump_midi1(dest, dest_port, event, atomic, hop); } + +/* return the UMP group-port number of the event; + * return -1 if groupless or non-UMP event + */ +int snd_seq_ump_group_port(const struct snd_seq_event *event) +{ + const struct snd_seq_ump_event *ump_ev = + (const struct snd_seq_ump_event *)event; + unsigned char type; + + if (!snd_seq_ev_is_ump(event)) + return -1; + type = ump_message_type(ump_ev->ump[0]); + if (ump_is_groupless_msg(type)) + return -1; + /* group-port number starts from 1 */ + return ump_message_group(ump_ev->ump[0]) + 1; +} diff --git a/sound/core/seq/seq_ump_convert.h b/sound/core/seq/seq_ump_convert.h index 6c146d803280..4abf0a7637d7 100644 --- a/sound/core/seq/seq_ump_convert.h +++ b/sound/core/seq/seq_ump_convert.h @@ -18,5 +18,6 @@ int snd_seq_deliver_to_ump(struct snd_seq_client *source, struct snd_seq_client_port *dest_port, struct snd_seq_event *event, int atomic, int hop); +int snd_seq_ump_group_port(const struct snd_seq_event *event); #endif /* __SEQ_UMP_CONVERT_H */ -- 2.51.0 From 1f93d877f09d987f08baedd50597aaaa72a37be4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 8 May 2025 21:12:07 +0300 Subject: [PATCH 09/16] ALSA/hda: intel-sdw-acpi: Correct sdw_intel_acpi_scan() function parameter The acpi_handle should be just a handle and not a pointer in sdw_intel_acpi_scan() parameter list. It is called with 'acpi_handle handle' as parameter and it is passing it to acpi_walk_namespace, which also expects acpi_handle and not acpi_handle* Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20250508181207.22113-1-peter.ujfalusi@linux.intel.com Signed-off-by: Takashi Iwai --- include/linux/soundwire/sdw_intel.h | 2 +- sound/hda/intel-sdw-acpi.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 493d9de4e472..dc6ebaee3d18 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -365,7 +365,7 @@ struct sdw_intel_res { * on e.g. which machine driver to select (I2S mode, HDaudio or * SoundWire). */ -int sdw_intel_acpi_scan(acpi_handle *parent_handle, +int sdw_intel_acpi_scan(acpi_handle parent_handle, struct sdw_intel_acpi_info *info); void sdw_intel_process_wakeen_event(struct sdw_intel_ctx *ctx); diff --git a/sound/hda/intel-sdw-acpi.c b/sound/hda/intel-sdw-acpi.c index 8686adaf4531..d3511135f7d3 100644 --- a/sound/hda/intel-sdw-acpi.c +++ b/sound/hda/intel-sdw-acpi.c @@ -177,7 +177,7 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, * sdw_intel_startup() is required for creation of devices and bus * startup */ -int sdw_intel_acpi_scan(acpi_handle *parent_handle, +int sdw_intel_acpi_scan(acpi_handle parent_handle, struct sdw_intel_acpi_info *info) { acpi_status status; -- 2.51.0 From dd33993a9721ab1dae38bd37c9f665987d554239 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 11 May 2025 16:11:45 +0200 Subject: [PATCH 10/16] ALSA: ump: Fix a typo of snd_ump_stream_msg_device_info s/devince/device/ It's used only internally, so no any behavior changes. Fixes: 37e0e14128e0 ("ALSA: ump: Support UMP Endpoint and Function Block parsing") Acked-by: Greg Kroah-Hartman Link: https://patch.msgid.link/20250511141147.10246-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- drivers/usb/gadget/function/f_midi2.c | 2 +- include/sound/ump_msg.h | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/usb/gadget/function/f_midi2.c b/drivers/usb/gadget/function/f_midi2.c index 12e866fb311d..0a800ba53816 100644 --- a/drivers/usb/gadget/function/f_midi2.c +++ b/drivers/usb/gadget/function/f_midi2.c @@ -475,7 +475,7 @@ static void reply_ump_stream_ep_info(struct f_midi2_ep *ep) /* reply a UMP EP device info */ static void reply_ump_stream_ep_device(struct f_midi2_ep *ep) { - struct snd_ump_stream_msg_devince_info rep = { + struct snd_ump_stream_msg_device_info rep = { .type = UMP_MSG_TYPE_STREAM, .status = UMP_STREAM_MSG_STATUS_DEVICE_INFO, .manufacture_id = ep->info.manufacturer, diff --git a/include/sound/ump_msg.h b/include/sound/ump_msg.h index 72f60ddfea75..9556b4755a1e 100644 --- a/include/sound/ump_msg.h +++ b/include/sound/ump_msg.h @@ -604,7 +604,7 @@ struct snd_ump_stream_msg_ep_info { } __packed; /* UMP Stream Message: Device Info Notification (128bit) */ -struct snd_ump_stream_msg_devince_info { +struct snd_ump_stream_msg_device_info { #ifdef __BIG_ENDIAN_BITFIELD /* 0 */ u32 type:4; @@ -754,7 +754,7 @@ struct snd_ump_stream_msg_fb_name { union snd_ump_stream_msg { struct snd_ump_stream_msg_ep_discovery ep_discovery; struct snd_ump_stream_msg_ep_info ep_info; - struct snd_ump_stream_msg_devince_info device_info; + struct snd_ump_stream_msg_device_info device_info; struct snd_ump_stream_msg_stream_cfg stream_cfg; struct snd_ump_stream_msg_fb_discovery fb_discovery; struct snd_ump_stream_msg_fb_info fb_info; -- 2.51.0 From 2b24eb060c2bb9ef79e1d3bcf633ba1bc95215d6 Mon Sep 17 00:00:00 2001 From: Christian Heusel Date: Mon, 12 May 2025 22:23:37 +0200 Subject: [PATCH 11/16] ALSA: usb-audio: Add sample rate quirk for Audioengine D1 A user reported on the Arch Linux Forums that their device is emitting the following message in the kernel journal, which is fixed by adding the quirk as submitted in this patch: > kernel: usb 1-2: current rate 8436480 is different from the runtime rate 48000 There also is an entry for this product line added long time ago. Their specific device has the following ID: $ lsusb | grep Audio Bus 001 Device 002: ID 1101:0003 EasyPass Industrial Co., Ltd Audioengine D1 Link: https://bbs.archlinux.org/viewtopic.php?id=305494 Fixes: 93f9d1a4ac593 ("ALSA: usb-audio: Apply sample rate quirk for Audioengine D1") Cc: stable@vger.kernel.org Signed-off-by: Christian Heusel Link: https://patch.msgid.link/20250512-audioengine-quirk-addition-v1-1-4c370af6eff7@heusel.eu Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9112313a9dbc..eb192834db68 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2250,6 +2250,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x0fd9, 0x0008, /* Hauppauge HVR-950Q */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x1101, 0x0003, /* Audioengine D1 */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16), DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */ -- 2.51.0 From 66e48ef6ef506c89ec1b3851c6f9f5f80b5835ff Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Tue, 13 May 2025 09:31:04 +0200 Subject: [PATCH 12/16] ALSA: sh: SND_AICA should depend on SH_DMA_API If CONFIG_SH_DMA_API=n: WARNING: unmet direct dependencies detected for G2_DMA Depends on [n]: SH_DREAMCAST [=y] && SH_DMA_API [=n] Selected by [y]: - SND_AICA [=y] && SOUND [=y] && SND [=y] && SND_SUPERH [=y] && SH_DREAMCAST [=y] SND_AICA selects G2_DMA. As the latter depends on SH_DMA_API, the former should depend on SH_DMA_API, too. Fixes: f477a538c14d07f8 ("sh: dma: fix kconfig dependency for G2_DMA") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202505131320.PzgTtl9H-lkp@intel.com/ Signed-off-by: Geert Uytterhoeven Link: https://patch.msgid.link/b90625f8a9078d0d304bafe862cbe3a3fab40082.1747121335.git.geert+renesas@glider.be Signed-off-by: Takashi Iwai --- sound/sh/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig index b75fbb3236a7..f5fa09d740b4 100644 --- a/sound/sh/Kconfig +++ b/sound/sh/Kconfig @@ -14,7 +14,7 @@ if SND_SUPERH config SND_AICA tristate "Dreamcast Yamaha AICA sound" - depends on SH_DREAMCAST + depends on SH_DREAMCAST && SH_DMA_API select SND_PCM select G2_DMA help -- 2.51.0 From 9e000f1b7f31684cc5927e034360b87ac7919593 Mon Sep 17 00:00:00 2001 From: Wentao Liang Date: Wed, 14 May 2025 17:24:44 +0800 Subject: [PATCH 13/16] ALSA: es1968: Add error handling for snd_pcm_hw_constraint_pow2() The function snd_es1968_capture_open() calls the function snd_pcm_hw_constraint_pow2(), but does not check its return value. A proper implementation can be found in snd_cx25821_pcm_open(). Add error handling for snd_pcm_hw_constraint_pow2() and propagate its error code. Fixes: b942cf815b57 ("[ALSA] es1968 - Fix stuttering capture") Cc: stable@vger.kernel.org # v2.6.22 Signed-off-by: Wentao Liang Link: https://patch.msgid.link/20250514092444.331-1-vulab@iscas.ac.cn Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index c6c018b40c69..4e0693f0ab0f 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1561,7 +1561,7 @@ static int snd_es1968_capture_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct es1968 *chip = snd_pcm_substream_chip(substream); struct esschan *es; - int apu1, apu2; + int err, apu1, apu2; apu1 = snd_es1968_alloc_apu_pair(chip, ESM_APU_PCM_CAPTURE); if (apu1 < 0) @@ -1605,7 +1605,9 @@ static int snd_es1968_capture_open(struct snd_pcm_substream *substream) runtime->hw = snd_es1968_capture; runtime->hw.buffer_bytes_max = runtime->hw.period_bytes_max = calc_available_memory_size(chip) - 1024; /* keep MIXBUF size */ - snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES); + err = snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES); + if (err < 0) + return err; spin_lock_irq(&chip->substream_lock); list_add(&es->list, &chip->substream_list); -- 2.51.0 From 7b9938a14460e8ec7649ca2e80ac0aae9815bf02 Mon Sep 17 00:00:00 2001 From: Nicolas Chauvet Date: Thu, 15 May 2025 12:21:32 +0200 Subject: [PATCH 14/16] ALSA: usb-audio: Add sample rate quirk for Microdia JP001 USB Camera Microdia JP001 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x84". This patch adds the USB ID to quirks.c and avoids those error messages. usb 7-4: New USB device found, idVendor=0c45, idProduct=636b, bcdDevice= 1.00 usb 7-4: New USB device strings: Mfr=2, Product=1, SerialNumber=3 usb 7-4: Product: JP001 usb 7-4: Manufacturer: JP001 usb 7-4: SerialNumber: JP001 usb 7-4: 3:1: cannot get freq at ep 0x84 Cc: Signed-off-by: Nicolas Chauvet Link: https://patch.msgid.link/20250515102132.73062-1-kwizart@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index eb192834db68..dbbc9eb935a4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2242,6 +2242,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x0c45, 0x6340, /* Sonix HD USB Camera */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x0c45, 0x636b, /* Microdia JP001 USB Camera */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x0d8c, 0x0014, /* USB Audio Device */ QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x0ecb, 0x205c, /* JBL Quantum610 Wireless */ -- 2.51.0 From 93a81ca0657758b607c3f4ba889ae806be9beb73 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 May 2025 10:08:16 +0200 Subject: [PATCH 15/16] ALSA: pcm: Fix race of buffer access at PCM OSS layer The PCM OSS layer tries to clear the buffer with the silence data at initialization (or reconfiguration) of a stream with the explicit call of snd_pcm_format_set_silence() with runtime->dma_area. But this may lead to a UAF because the accessed runtime->dma_area might be freed concurrently, as it's performed outside the PCM ops. For avoiding it, move the code into the PCM core and perform it inside the buffer access lock, so that it won't be changed during the operation. Reported-by: syzbot+32d4647f551007595173@syzkaller.appspotmail.com Closes: https://lore.kernel.org/68164d8e.050a0220.11da1b.0019.GAE@google.com Cc: Link: https://patch.msgid.link/20250516080817.20068-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 ++ sound/core/oss/pcm_oss.c | 3 +-- sound/core/pcm_native.c | 11 +++++++++++ 3 files changed, 14 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 8becb4504887..8582d22f3818 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1404,6 +1404,8 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif +void snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime); + /** * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer * @dma: DMA number diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 4683b9139c56..4ecb17bd5436 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1074,8 +1074,7 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) runtime->oss.params = 0; runtime->oss.prepare = 1; runtime->oss.buffer_used = 0; - if (runtime->dma_area) - snd_pcm_format_set_silence(runtime->format, runtime->dma_area, bytes_to_samples(runtime, runtime->dma_bytes)); + snd_pcm_runtime_buffer_set_silence(runtime); runtime->oss.period_frames = snd_pcm_alsa_frames(substream, oss_period_size); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 6c2b6a62d9d2..853ac5bb33ff 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -723,6 +723,17 @@ static void snd_pcm_buffer_access_unlock(struct snd_pcm_runtime *runtime) atomic_inc(&runtime->buffer_accessing); } +/* fill the PCM buffer with the current silence format; called from pcm_oss.c */ +void snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime) +{ + snd_pcm_buffer_access_lock(runtime); + if (runtime->dma_area) + snd_pcm_format_set_silence(runtime->format, runtime->dma_area, + bytes_to_samples(runtime, runtime->dma_bytes)); + snd_pcm_buffer_access_unlock(runtime); +} +EXPORT_SYMBOL_GPL(snd_pcm_runtime_buffer_set_silence); + #if IS_ENABLED(CONFIG_SND_PCM_OSS) #define is_oss_stream(substream) ((substream)->oss.oss) #else -- 2.51.0 From 5ad8a4ddc45048bc2fe23b75357b6bf185db004f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 16 May 2025 14:53:37 +0800 Subject: [PATCH 16/16] ALSA: hda/realtek - restore auto-mute mode for Dell Chrome platform This board need to shutdown Class-D amp to avoid EMI issue. Restore the Auto-Mute mode item will off pin control when Auto-mute mode was enable. Signed-off-by: Kailang Yang Links: https://lore.kernel.org/ee8bbe5236464c369719d96269ba8ef8@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a2b09e4a7d5..dcfaddc3ae13 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6830,7 +6830,10 @@ static void alc256_fixup_chromebook(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: - spec->gen.suppress_auto_mute = 1; + if (codec->core.subsystem_id == 0x10280d76) + spec->gen.suppress_auto_mute = 0; + else + spec->gen.suppress_auto_mute = 1; spec->gen.suppress_auto_mic = 1; spec->en_3kpull_low = false; break; -- 2.51.0