From 579cd64b9df8a60284ec3422be919c362de40e41 Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Sat, 8 Feb 2025 00:54:35 +0000 Subject: [PATCH 01/14] ASoC: tas2770: Fix volume scale The scale starts at -100dB, not -128dB. Signed-off-by: Hector Martin Signed-off-by: Mark Brown Link: https://patch.msgid.link/20250208-asoc-tas2770-v1-1-cf50ff1d59a3@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas2770.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 9f93b230652a..863c3f672ba9 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -506,7 +506,7 @@ static int tas2770_codec_probe(struct snd_soc_component *component) } static DECLARE_TLV_DB_SCALE(tas2770_digital_tlv, 1100, 50, 0); -static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -12750, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -10050, 50, 0); static const struct snd_kcontrol_new tas2770_snd_controls[] = { SOC_SINGLE_TLV("Speaker Playback Volume", TAS2770_PLAY_CFG_REG2, From a3bdd8f5c2217e1cb35db02c2eed36ea20fb50f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2025 12:40:24 +0100 Subject: [PATCH 02/14] ALSA: usb-audio: Avoid dropping MIDI events at closing multiple ports We fixed the UAF issue in USB MIDI code by canceling the pending work at closing each MIDI output device in the commit below. However, this assumed that it's the only one that is tied with the endpoint, and it resulted in unexpected data truncations when multiple devices are assigned to a single endpoint and opened simultaneously. For addressing the unexpected MIDI message drops, simply replace cancel_work_sync() with flush_work(). The drain callback should have been already invoked before the close callback, hence the port->active flag must be already cleared. So this just assures that the pending work is finished before freeing the resources. Fixes: 0125de38122f ("ALSA: usb-audio: Cancel pending work at closing a MIDI substream") Reported-and-tested-by: John Keeping Closes: https://lore.kernel.org/20250217111647.3368132-1-jkeeping@inmusicbrands.com Link: https://patch.msgid.link/20250218114024.23125-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 737dd00e97b1..779d97d31f17 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1145,7 +1145,7 @@ static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { struct usbmidi_out_port *port = substream->runtime->private_data; - cancel_work_sync(&port->ep->work); + flush_work(&port->ep->work); return substream_open(substream, 0, 0); } From a3f172359e22b2c11b750d23560481a55bf86af1 Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Tue, 18 Feb 2025 18:35:35 +1000 Subject: [PATCH 03/14] ASoC: tas2764: Fix power control mask Reviewed-by: Neal Gompa Signed-off-by: Hector Martin Signed-off-by: James Calligeros Link: https://patch.msgid.link/20250218-apple-codec-changes-v2-1-932760fd7e07@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2764.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h index 168af772a898..d13ecae9c9c2 100644 --- a/sound/soc/codecs/tas2764.h +++ b/sound/soc/codecs/tas2764.h @@ -25,7 +25,7 @@ /* Power Control */ #define TAS2764_PWR_CTRL TAS2764_REG(0X0, 0x02) -#define TAS2764_PWR_CTRL_MASK GENMASK(1, 0) +#define TAS2764_PWR_CTRL_MASK GENMASK(2, 0) #define TAS2764_PWR_CTRL_ACTIVE 0x0 #define TAS2764_PWR_CTRL_MUTE BIT(0) #define TAS2764_PWR_CTRL_SHUTDOWN BIT(1) From f5468beeab1b1adfc63c2717b1f29ef3f49a5fab Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Tue, 18 Feb 2025 18:36:02 +1000 Subject: [PATCH 04/14] ASoC: tas2764: Set the SDOUT polarity correctly TX launch polarity needs to be the opposite of RX capture polarity, to generate the right bit slot alignment. Reviewed-by: Neal Gompa Signed-off-by: Hector Martin Signed-off-by: James Calligeros Link: https://patch.msgid.link/20250218-apple-codec-changes-v2-28-932760fd7e07@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2764.c | 10 +++++++++- sound/soc/codecs/tas2764.h | 6 ++++++ 2 files changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index d482cd194c08..58315eab492a 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -365,7 +365,7 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component); - u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0; + u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0, asi_cfg_4 = 0; int ret; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -374,12 +374,14 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) fallthrough; case SND_SOC_DAIFMT_NB_NF: asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING; + asi_cfg_4 = TAS2764_TDM_CFG4_TX_FALLING; break; case SND_SOC_DAIFMT_IB_IF: asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START; fallthrough; case SND_SOC_DAIFMT_IB_NF: asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING; + asi_cfg_4 = TAS2764_TDM_CFG4_TX_RISING; break; } @@ -389,6 +391,12 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) if (ret < 0) return ret; + ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG4, + TAS2764_TDM_CFG4_TX_MASK, + asi_cfg_4); + if (ret < 0) + return ret; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START; diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h index d13ecae9c9c2..9490f2686e38 100644 --- a/sound/soc/codecs/tas2764.h +++ b/sound/soc/codecs/tas2764.h @@ -79,6 +79,12 @@ #define TAS2764_TDM_CFG3_RXS_SHIFT 0x4 #define TAS2764_TDM_CFG3_MASK GENMASK(3, 0) +/* TDM Configuration Reg4 */ +#define TAS2764_TDM_CFG4 TAS2764_REG(0X0, 0x0d) +#define TAS2764_TDM_CFG4_TX_MASK BIT(0) +#define TAS2764_TDM_CFG4_TX_RISING 0x0 +#define TAS2764_TDM_CFG4_TX_FALLING BIT(0) + /* TDM Configuration Reg5 */ #define TAS2764_TDM_CFG5 TAS2764_REG(0X0, 0x0e) #define TAS2764_TDM_CFG5_VSNS_MASK BIT(6) From 9af3b4f2d879da01192d6168e6c651e7fb5b652d Mon Sep 17 00:00:00 2001 From: Dmitry Panchenko Date: Thu, 20 Feb 2025 18:15:37 +0200 Subject: [PATCH 05/14] ALSA: usb-audio: Re-add sample rate quirk for Pioneer DJM-900NXS2 Re-add the sample-rate quirk for the Pioneer DJM-900NXS2. This device does not work without setting sample-rate. Signed-off-by: Dmitry Panchenko Cc: Link: https://patch.msgid.link/20250220161540.3624660-1-dmitry@d-systems.ee Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a97efb7b131e..09210fb4ac60 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1868,6 +1868,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x2b73, 0x000a): /* Pioneer DJM-900NXS2 */ case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ pioneer_djm_set_format_quirk(subs, 0x0082); break; From 5b0c02f9b8acf2a791e531bbc09acae2d51f4f9b Mon Sep 17 00:00:00 2001 From: Nicolas Frattaroli Date: Sat, 22 Feb 2025 20:39:57 +0100 Subject: [PATCH 06/14] ASoC: es8328: fix route from DAC to output The ES8328 codec driver, which is also used for the ES8388 chip that appears to have an identical register map, claims that the output can either take the route from DAC->Mixer->Output or through DAC->Output directly. To the best of what I could find, this is not true, and creates problems. Without DACCONTROL17 bit index 7 set for the left channel, as well as DACCONTROL20 bit index 7 set for the right channel, I cannot get any analog audio out on Left Out 2 and Right Out 2 respectively, despite the DAPM routes claiming that this should be possible. Furthermore, the same is the case for Left Out 1 and Right Out 1, showing that those two don't have a direct route from DAC to output bypassing the mixer either. Those control bits toggle whether the DACs are fed (stale bread?) into their respective mixers. If one "unmutes" the mixer controls in alsamixer, then sure, the audio output works, but if it doesn't work without the mixer being fed the DAC input then evidently it's not a direct output from the DAC. ES8328/ES8388 are seemingly not alone in this. ES8323, which uses a separate driver for what appears to be a very similar register map, simply flips those two bits on in its probe function, and then pretends there is no power management whatsoever for the individual controls. Fair enough. My theory as to why nobody has noticed this up to this point is that everyone just assumes it's their fault when they had to unmute an additional control in ALSA. Fix this in the es8328 driver by removing the erroneous direct route, then get rid of the playback switch controls and have those bits tied to the mixer's widget instead, which until now had no register to play with. Fixes: 567e4f98922c ("ASoC: add es8328 codec driver") Signed-off-by: Nicolas Frattaroli Link: https://patch.msgid.link/20250222-es8328-route-bludgeoning-v1-1-99bfb7fb22d9@collabora.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8328.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index f3c97da798dc..76159c45e6b5 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -233,7 +233,6 @@ static const struct snd_kcontrol_new es8328_right_line_controls = /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0), SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0), SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0), SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0), @@ -243,7 +242,6 @@ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { static const struct snd_kcontrol_new es8328_right_mixer_controls[] = { SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0), SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0), - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0), SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0), }; @@ -336,10 +334,10 @@ static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = { SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER, ES8328_DACPOWER_LDAC_OFF, 1), - SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_MIXER("Left Mixer", ES8328_DACCONTROL17, 7, 0, &es8328_left_mixer_controls[0], ARRAY_SIZE(es8328_left_mixer_controls)), - SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_MIXER("Right Mixer", ES8328_DACCONTROL20, 7, 0, &es8328_right_mixer_controls[0], ARRAY_SIZE(es8328_right_mixer_controls)), @@ -418,19 +416,14 @@ static const struct snd_soc_dapm_route es8328_dapm_routes[] = { { "Right Line Mux", "PGA", "Right PGA Mux" }, { "Right Line Mux", "Differential", "Differential Mux" }, - { "Left Out 1", NULL, "Left DAC" }, - { "Right Out 1", NULL, "Right DAC" }, - { "Left Out 2", NULL, "Left DAC" }, - { "Right Out 2", NULL, "Right DAC" }, - - { "Left Mixer", "Playback Switch", "Left DAC" }, + { "Left Mixer", NULL, "Left DAC" }, { "Left Mixer", "Left Bypass Switch", "Left Line Mux" }, { "Left Mixer", "Right Playback Switch", "Right DAC" }, { "Left Mixer", "Right Bypass Switch", "Right Line Mux" }, { "Right Mixer", "Left Playback Switch", "Left DAC" }, { "Right Mixer", "Left Bypass Switch", "Left Line Mux" }, - { "Right Mixer", "Playback Switch", "Right DAC" }, + { "Right Mixer", NULL, "Right DAC" }, { "Right Mixer", "Right Bypass Switch", "Right Line Mux" }, { "DAC DIG", NULL, "DAC STM" }, From 0da83ab025bc45e9742e87c2cce19bff423377c8 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Mon, 17 Feb 2025 10:04:37 +0900 Subject: [PATCH 07/14] ASoC: fsl: Rename stream name of SAI DAI driver If stream names of DAI driver are duplicated there'll be warnings when machine driver tries to add widgets on a route: [ 8.831335] fsl-asoc-card sound-wm8960: ASoC: sink widget CPU-Playback overwritten [ 8.839917] fsl-asoc-card sound-wm8960: ASoC: source widget CPU-Capture overwritten Use different stream names to avoid such warnings. DAI names in AUDMIX are also updated accordingly. Fixes: 15c958390460 ("ASoC: fsl_sai: Add separate DAI for transmitter and receiver") Signed-off-by: Chancel Liu Acked-by: Shengjiu Wang Link: https://patch.msgid.link/20250217010437.258621-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 6 +++--- sound/soc/fsl/imx-audmix.c | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c4eb87c5d39e..9f33dd11d47f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -994,10 +994,10 @@ static struct snd_soc_dai_driver fsl_sai_dai_template[] = { { .name = "sai-tx", .playback = { - .stream_name = "CPU-Playback", + .stream_name = "SAI-Playback", .channels_min = 1, .channels_max = 32, - .rate_min = 8000, + .rate_min = 8000, .rate_max = 2822400, .rates = SNDRV_PCM_RATE_KNOT, .formats = FSL_SAI_FORMATS, @@ -1007,7 +1007,7 @@ static struct snd_soc_dai_driver fsl_sai_dai_template[] = { { .name = "sai-rx", .capture = { - .stream_name = "CPU-Capture", + .stream_name = "SAI-Capture", .channels_min = 1, .channels_max = 32, .rate_min = 8000, diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 50ecc5f51100..dac5d4ddacd6 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -119,8 +119,8 @@ static const struct snd_soc_ops imx_audmix_be_ops = { static const char *name[][3] = { {"HiFi-AUDMIX-FE-0", "HiFi-AUDMIX-FE-1", "HiFi-AUDMIX-FE-2"}, {"sai-tx", "sai-tx", "sai-rx"}, - {"AUDMIX-Playback-0", "AUDMIX-Playback-1", "CPU-Capture"}, - {"CPU-Playback", "CPU-Playback", "AUDMIX-Capture-0"}, + {"AUDMIX-Playback-0", "AUDMIX-Playback-1", "SAI-Capture"}, + {"SAI-Playback", "SAI-Playback", "AUDMIX-Capture-0"}, }; static int imx_audmix_probe(struct platform_device *pdev) From d31babd7e304d3b800d36ff74be6739405b985f2 Mon Sep 17 00:00:00 2001 From: Nicolas Frattaroli Date: Fri, 21 Feb 2025 21:39:32 +0100 Subject: [PATCH 08/14] ASoC: dapm-graph: set fill colour of turned on nodes Some tools like KGraphViewer interpret the "ON" nodes not having an explicitly set fill colour as them being entirely black, which obscures the text on them and looks funny. In fact, I thought they were off for the longest time. Comparing to the output of the `dot` tool, I assume they are supposed to be white. Instead of speclawyering over who's in the wrong and must immediately atone for their wickedness at the altar of RFC2119, just be explicit about it, set the fillcolor to white, and nobody gets confused. Signed-off-by: Nicolas Frattaroli Tested-by: Luca Ceresoli Reviewed-by: Luca Ceresoli Link: https://patch.msgid.link/20250221-dapm-graph-node-colour-v1-1-514ed0aa7069@collabora.com Signed-off-by: Mark Brown --- tools/sound/dapm-graph | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/tools/sound/dapm-graph b/tools/sound/dapm-graph index f14bdfedee8f..b6196ee5065a 100755 --- a/tools/sound/dapm-graph +++ b/tools/sound/dapm-graph @@ -10,7 +10,7 @@ set -eu STYLE_COMPONENT_ON="color=dodgerblue;style=bold" STYLE_COMPONENT_OFF="color=gray40;style=filled;fillcolor=gray90" -STYLE_NODE_ON="shape=box,style=bold,color=green4" +STYLE_NODE_ON="shape=box,style=bold,color=green4,fillcolor=white" STYLE_NODE_OFF="shape=box,style=filled,color=gray30,fillcolor=gray95" # Print usage and exit From 56a677293509b2a0d39ac8d02b583c1ab1fe4d94 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 25 Feb 2025 17:37:15 +0800 Subject: [PATCH 09/14] ASoC: SOF: Intel: don't check number of sdw links when set dmic_fixup MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Currently, we assume that the PCH DMIC pins are pin-muxed with SoundWire links. However, we do see a HW design that use PCH DMIC along with 3 SoundWire links. Remove the check now. With this change the PCM DMIC will be presented if it is reported by the BIOS irrespective of whether there are SDW links present or not. Signed-off-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://patch.msgid.link/20250225093716.67240-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 18 ++---------------- 1 file changed, 2 insertions(+), 16 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index be689f6e10c8..a1ccd95da8bb 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1312,22 +1312,8 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) /* report to machine driver if any DMICs are found */ mach->mach_params.dmic_num = check_dmic_num(sdev); - if (sdw_mach_found) { - /* - * DMICs use up to 4 pins and are typically pin-muxed with SoundWire - * link 2 and 3, or link 1 and 2, thus we only try to enable dmics - * if all conditions are true: - * a) 2 or fewer links are used by SoundWire - * b) the NHLT table reports the presence of microphones - */ - if (hweight_long(mach->link_mask) <= 2) - dmic_fixup = true; - else - mach->mach_params.dmic_num = 0; - } else { - if (mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER) - dmic_fixup = true; - } + if (sdw_mach_found || mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER) + dmic_fixup = true; if (tplg_fixup && dmic_fixup && From 4bcef04ca67fa8c46591a211885bb9c37f25dbbd Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 25 Feb 2025 17:37:16 +0800 Subject: [PATCH 10/14] ASoC: Intel: sof_sdw: warn both sdw and pch dmic are used MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Typically, SoundWire MIC and PCH DMIC will not coexist. However, we may want to use both of them in some special cases. Add a warning to let users know that SoundWire MIC and PCH DMIC are both present and they could overwrite it with kernel params. Signed-off-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://patch.msgid.link/20250225093716.67240-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 203b07d4d833..c13064c77726 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -803,7 +803,9 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *be_id, struct snd_soc_codec_conf **codec_conf) { struct device *dev = card->dev; + struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev); struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; struct asoc_sdw_endpoint *sof_end; int stream; @@ -900,6 +902,11 @@ static int create_sdw_dailink(struct snd_soc_card *card, codecs[j].name = sof_end->codec_name; codecs[j].dai_name = sof_end->dai_info->dai_name; + if (sof_end->dai_info->dai_type == SOC_SDW_DAI_TYPE_MIC && + mach_params->dmic_num > 0) { + dev_warn(dev, + "Both SDW DMIC and PCH DMIC are present, if incorrect, please set kernel params snd_sof_intel_hda_generic dmic_num=0 to disable PCH DMIC\n"); + } j++; } From fe08b7d5085a9774abc30c26d5aebc5b9cdd6091 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 25 Feb 2025 13:18:42 +0000 Subject: [PATCH 11/14] firmware: cs_dsp: Remove async regmap writes Change calls to async regmap write functions to use the normal blocking writes so that the cs35l56 driver can use spi_bus_lock() to gain exclusive access to the SPI bus. As this is part of a fix, it makes only the minimal change to swap the functions to the blocking equivalents. There's no need to risk reworking the buffer allocation logic that is now partially redundant. The async writes are a 12-year-old workaround for inefficiency of synchronous writes in the SPI subsystem. The SPI subsystem has since been changed to avoid the overheads, so this workaround should not be necessary. The cs35l56 driver needs to use spi_bus_lock() prevent bus activity while it is soft-resetting the cs35l56. But spi_bus_lock() is incompatible with spi_async() calls, which will fail with -EBUSY. Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file") Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20250225131843.113752-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- drivers/firmware/cirrus/cs_dsp.c | 24 ++++++------------------ 1 file changed, 6 insertions(+), 18 deletions(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 5365e9a43000..42433c19eb30 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -1609,8 +1609,8 @@ static int cs_dsp_load(struct cs_dsp *dsp, const struct firmware *firmware, goto out_fw; } - ret = regmap_raw_write_async(regmap, reg, buf->buf, - le32_to_cpu(region->len)); + ret = regmap_raw_write(regmap, reg, buf->buf, + le32_to_cpu(region->len)); if (ret != 0) { cs_dsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", @@ -1625,12 +1625,6 @@ static int cs_dsp_load(struct cs_dsp *dsp, const struct firmware *firmware, regions++; } - ret = regmap_async_complete(regmap); - if (ret != 0) { - cs_dsp_err(dsp, "Failed to complete async write: %d\n", ret); - goto out_fw; - } - if (pos > firmware->size) cs_dsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, regions, pos - firmware->size); @@ -1638,7 +1632,6 @@ static int cs_dsp_load(struct cs_dsp *dsp, const struct firmware *firmware, cs_dsp_debugfs_save_wmfwname(dsp, file); out_fw: - regmap_async_complete(regmap); cs_dsp_buf_free(&buf_list); if (ret == -EOVERFLOW) @@ -2326,8 +2319,8 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware cs_dsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", file, blocks, le32_to_cpu(blk->len), reg); - ret = regmap_raw_write_async(regmap, reg, buf->buf, - le32_to_cpu(blk->len)); + ret = regmap_raw_write(regmap, reg, buf->buf, + le32_to_cpu(blk->len)); if (ret != 0) { cs_dsp_err(dsp, "%s.%d: Failed to write to %x in %s: %d\n", @@ -2339,10 +2332,6 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware blocks++; } - ret = regmap_async_complete(regmap); - if (ret != 0) - cs_dsp_err(dsp, "Failed to complete async write: %d\n", ret); - if (pos > firmware->size) cs_dsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, blocks, pos - firmware->size); @@ -2350,7 +2339,6 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware cs_dsp_debugfs_save_binname(dsp, file); out_fw: - regmap_async_complete(regmap); cs_dsp_buf_free(&buf_list); if (ret == -EOVERFLOW) @@ -2561,8 +2549,8 @@ static int cs_dsp_adsp2_enable_core(struct cs_dsp *dsp) { int ret; - ret = regmap_update_bits_async(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA, ADSP2_SYS_ENA); + ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA, ADSP2_SYS_ENA); if (ret != 0) return ret; From 769c1b79295c38d60fde4c0a8f5f31e01360c54f Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 25 Feb 2025 13:18:43 +0000 Subject: [PATCH 12/14] ASoC: cs35l56: Prevent races when soft-resetting using SPI control When SPI is used for control, the driver must hold the SPI bus lock while issuing the sequence of writes to perform a soft reset. >From the time the driver writes the SYSTEM_RESET command until the driver does a write to terminate the reset, there must not be any activity on the SPI bus lines. If there is any SPI activity during the soft-reset, another soft-reset will be triggered. The state of the SPI chip select is irrelevant. A repeated soft-reset does not in itself cause any problems, and it is not an infinite loop. The problem is a race between these resets and the driver polling for boot completion. There is a time window between soft resets where the driver could read HALO_STATE as 2 (fully booted) while the chip is actually soft-resetting. Although this window is small, it is long enough that it is possible to hit it in normal operation. To prevent this race and ensure the chip really is fully booted, the driver calls spi_bus_lock() to prevent other activity while resetting. It then issues the SYSTEM_RESET mailbox command. After allowing sufficient time for reset to take effect, the driver issues a PING mailbox command, which will force completion of the full soft-reset sequence. The SPI bus lock can then be released. The mailbox is checked for any boot or wakeup response from the firmware, before the value in HALO_STATE will be trusted. This does not affect SoundWire or I2C control. Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file") Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20250225131843.113752-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 31 ++++++++++++ sound/pci/hda/cs35l56_hda_spi.c | 3 ++ sound/soc/codecs/cs35l56-shared.c | 80 +++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l56-spi.c | 3 ++ 4 files changed, 117 insertions(+) diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index 3dc7a1551ac3..5d653a3491d0 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -12,6 +12,7 @@ #include #include #include +#include #include #define CS35L56_DEVID 0x0000000 @@ -61,6 +62,7 @@ #define CS35L56_IRQ1_MASK_8 0x000E0AC #define CS35L56_IRQ1_MASK_18 0x000E0D4 #define CS35L56_IRQ1_MASK_20 0x000E0DC +#define CS35L56_DSP_MBOX_1_RAW 0x0011000 #define CS35L56_DSP_VIRTUAL1_MBOX_1 0x0011020 #define CS35L56_DSP_VIRTUAL1_MBOX_2 0x0011024 #define CS35L56_DSP_VIRTUAL1_MBOX_3 0x0011028 @@ -224,6 +226,7 @@ #define CS35L56_HALO_STATE_SHUTDOWN 1 #define CS35L56_HALO_STATE_BOOT_DONE 2 +#define CS35L56_MBOX_CMD_PING 0x0A000000 #define CS35L56_MBOX_CMD_AUDIO_PLAY 0x0B000001 #define CS35L56_MBOX_CMD_AUDIO_PAUSE 0x0B000002 #define CS35L56_MBOX_CMD_AUDIO_REINIT 0x0B000003 @@ -254,6 +257,16 @@ #define CS35L56_NUM_BULK_SUPPLIES 3 #define CS35L56_NUM_DSP_REGIONS 5 +/* Additional margin for SYSTEM_RESET to control port ready on SPI */ +#define CS35L56_SPI_RESET_TO_PORT_READY_US (CS35L56_CONTROL_PORT_READY_US + 2500) + +struct cs35l56_spi_payload { + __be32 addr; + __be16 pad; + __be32 value; +} __packed; +static_assert(sizeof(struct cs35l56_spi_payload) == 10); + struct cs35l56_base { struct device *dev; struct regmap *regmap; @@ -269,6 +282,7 @@ struct cs35l56_base { s8 cal_index; struct cirrus_amp_cal_data cal_data; struct gpio_desc *reset_gpio; + struct cs35l56_spi_payload *spi_payload_buf; }; static inline bool cs35l56_is_otp_register(unsigned int reg) @@ -276,6 +290,23 @@ static inline bool cs35l56_is_otp_register(unsigned int reg) return (reg >> 16) == 3; } +static inline int cs35l56_init_config_for_spi(struct cs35l56_base *cs35l56, + struct spi_device *spi) +{ + cs35l56->spi_payload_buf = devm_kzalloc(&spi->dev, + sizeof(*cs35l56->spi_payload_buf), + GFP_KERNEL | GFP_DMA); + if (!cs35l56->spi_payload_buf) + return -ENOMEM; + + return 0; +} + +static inline bool cs35l56_is_spi(struct cs35l56_base *cs35l56) +{ + return IS_ENABLED(CONFIG_SPI_MASTER) && !!cs35l56->spi_payload_buf; +} + extern const struct regmap_config cs35l56_regmap_i2c; extern const struct regmap_config cs35l56_regmap_spi; extern const struct regmap_config cs35l56_regmap_sdw; diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index d4ee5bb7c486..903578466905 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -22,6 +22,9 @@ static int cs35l56_hda_spi_probe(struct spi_device *spi) return -ENOMEM; cs35l56->base.dev = &spi->dev; + ret = cs35l56_init_config_for_spi(&cs35l56->base, spi); + if (ret) + return ret; #ifdef CS35L56_WAKE_HOLD_TIME_US cs35l56->base.can_hibernate = true; diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e0ed4fc11155..e28bfefa72f3 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -10,6 +10,7 @@ #include #include #include +#include #include #include @@ -303,6 +304,79 @@ void cs35l56_wait_min_reset_pulse(void) } EXPORT_SYMBOL_NS_GPL(cs35l56_wait_min_reset_pulse, "SND_SOC_CS35L56_SHARED"); +static const struct { + u32 addr; + u32 value; +} cs35l56_spi_system_reset_stages[] = { + { .addr = CS35L56_DSP_VIRTUAL1_MBOX_1, .value = CS35L56_MBOX_CMD_SYSTEM_RESET }, + /* The next write is necessary to delimit the soft reset */ + { .addr = CS35L56_DSP_MBOX_1_RAW, .value = CS35L56_MBOX_CMD_PING }, +}; + +static void cs35l56_spi_issue_bus_locked_reset(struct cs35l56_base *cs35l56_base, + struct spi_device *spi) +{ + struct cs35l56_spi_payload *buf = cs35l56_base->spi_payload_buf; + struct spi_transfer t = { + .tx_buf = buf, + .len = sizeof(*buf), + }; + struct spi_message m; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(cs35l56_spi_system_reset_stages); i++) { + buf->addr = cpu_to_be32(cs35l56_spi_system_reset_stages[i].addr); + buf->value = cpu_to_be32(cs35l56_spi_system_reset_stages[i].value); + spi_message_init_with_transfers(&m, &t, 1); + ret = spi_sync_locked(spi, &m); + if (ret) + dev_warn(cs35l56_base->dev, "spi_sync failed: %d\n", ret); + + usleep_range(CS35L56_SPI_RESET_TO_PORT_READY_US, + 2 * CS35L56_SPI_RESET_TO_PORT_READY_US); + } +} + +static void cs35l56_spi_system_reset(struct cs35l56_base *cs35l56_base) +{ + struct spi_device *spi = to_spi_device(cs35l56_base->dev); + unsigned int val; + int read_ret, ret; + + /* + * There must not be any other SPI bus activity while the amp is + * soft-resetting. + */ + ret = spi_bus_lock(spi->controller); + if (ret) { + dev_warn(cs35l56_base->dev, "spi_bus_lock failed: %d\n", ret); + return; + } + + cs35l56_spi_issue_bus_locked_reset(cs35l56_base, spi); + spi_bus_unlock(spi->controller); + + /* + * Check firmware boot by testing for a response in MBOX_2. + * HALO_STATE cannot be trusted yet because the reset sequence + * can leave it with stale state. But MBOX is reset. + * The regmap must remain in cache-only until the chip has + * booted, so use a bypassed read. + */ + ret = read_poll_timeout(regmap_read_bypassed, read_ret, + (val > 0) && (val < 0xffffffff), + CS35L56_HALO_STATE_POLL_US, + CS35L56_HALO_STATE_TIMEOUT_US, + false, + cs35l56_base->regmap, + CS35L56_DSP_VIRTUAL1_MBOX_2, + &val); + if (ret) { + dev_err(cs35l56_base->dev, "SPI reboot timed out(%d): MBOX2=%#x\n", + read_ret, val); + } +} + static const struct reg_sequence cs35l56_system_reset_seq[] = { REG_SEQ0(CS35L56_DSP1_HALO_STATE, 0), REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_SYSTEM_RESET), @@ -315,6 +389,12 @@ void cs35l56_system_reset(struct cs35l56_base *cs35l56_base, bool is_soundwire) * accesses other than the controlled system reset sequence below. */ regcache_cache_only(cs35l56_base->regmap, true); + + if (cs35l56_is_spi(cs35l56_base)) { + cs35l56_spi_system_reset(cs35l56_base); + return; + } + regmap_multi_reg_write_bypassed(cs35l56_base->regmap, cs35l56_system_reset_seq, ARRAY_SIZE(cs35l56_system_reset_seq)); diff --git a/sound/soc/codecs/cs35l56-spi.c b/sound/soc/codecs/cs35l56-spi.c index c101134e8532..ca6c03a8766d 100644 --- a/sound/soc/codecs/cs35l56-spi.c +++ b/sound/soc/codecs/cs35l56-spi.c @@ -33,6 +33,9 @@ static int cs35l56_spi_probe(struct spi_device *spi) cs35l56->base.dev = &spi->dev; cs35l56->base.can_hibernate = true; + ret = cs35l56_init_config_for_spi(&cs35l56->base, spi); + if (ret) + return ret; ret = cs35l56_common_probe(cs35l56); if (ret != 0) From 9e7c6779e3530bbdd465214afcd13f19c33e51a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Feb 2025 16:45:32 +0100 Subject: [PATCH 13/14] ALSA: hda/realtek: Fix wrong mic setup for ASUS VivoBook 15 ASUS VivoBook 15 with SSID 1043:1460 took an incorrect quirk via the pin pattern matching for ASUS (ALC256_FIXUP_ASUS_MIC), resulting in the two built-in mic pins (0x13 and 0x1b). This had worked without problems casually in the past because the right pin (0x1b) was picked up as the primary device. But since we fixed the pin enumeration for other bugs, the bogus one (0x13) is picked up as the primary device, hence the bug surfaced now. For addressing the regression, this patch explicitly specifies the quirk entry with ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, which sets up only the headset mic pin. Fixes: 3b4309546b48 ("ALSA: hda: Fix headset detection failure due to unstable sort") Closes: https://bugzilla.kernel.org/show_bug.cgi?id=219807 Link: https://patch.msgid.link/20250225154540.13543-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- 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 224616fbec4f..e5c80d4be535 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10623,6 +10623,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650PY/PZ/PV/PU/PYV/PZV/PIV/PVV", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1460, "Asus VivoBook 15", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1463, "Asus GA402X/GA402N", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604VI/VC/VE/VG/VJ/VQ/VU/VV/VY/VZ", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603VQ/VU/VV/VJ/VI", ALC285_FIXUP_ASUS_HEADSET_MIC), From c6557ccf8094ce2e1142c6e49cd47f5d5e2933a8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Adrien=20Verg=C3=A9?= Date: Wed, 26 Feb 2025 14:55:15 +0100 Subject: [PATCH 14/14] ALSA: hda/realtek: Fix microphone regression on ASUS N705UD MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This fixes a regression introduced a few weeks ago in stable kernels 6.12.14 and 6.13.3. The internal microphone on ASUS Vivobook N705UD / X705UD laptops is broken: the microphone appears in userspace (e.g. Gnome settings) but no sound is detected. I bisected it to commit 3b4309546b48 ("ALSA: hda: Fix headset detection failure due to unstable sort"). I figured out the cause: 1. The initial pins enabled for the ALC256 driver are: cfg->inputs == { { pin=0x19, type=AUTO_PIN_MIC, is_headset_mic=1, is_headphone_mic=0, has_boost_on_pin=1 }, { pin=0x1a, type=AUTO_PIN_MIC, is_headset_mic=0, is_headphone_mic=0, has_boost_on_pin=1 } } 2. Since 2017 and commits c1732ede5e8 ("ALSA: hda/realtek - Fix headset and mic on several ASUS laptops with ALC256") and 28e8af8a163 ("ALSA: hda/realtek: Fix mic and headset jack sense on ASUS X705UD"), the quirk ALC256_FIXUP_ASUS_MIC is also applied to ASUS X705UD / N705UD laptops. This added another internal microphone on pin 0x13: cfg->inputs == { { pin=0x13, type=AUTO_PIN_MIC, is_headset_mic=0, is_headphone_mic=0, has_boost_on_pin=1 }, { pin=0x19, type=AUTO_PIN_MIC, is_headset_mic=1, is_headphone_mic=0, has_boost_on_pin=1 }, { pin=0x1a, type=AUTO_PIN_MIC, is_headset_mic=0, is_headphone_mic=0, has_boost_on_pin=1 } } I don't know what this pin 0x13 corresponds to. To the best of my knowledge, these laptops have only one internal microphone. 3. Before 2025 and commit 3b4309546b48 ("ALSA: hda: Fix headset detection failure due to unstable sort"), the sort function would let the microphone of pin 0x1a (the working one) *before* the microphone of pin 0x13 (the phantom one). 4. After this commit 3b4309546b48, the fixed sort function puts the working microphone (pin 0x1a) *after* the phantom one (pin 0x13). As a result, no sound is detected anymore. It looks like the quirk ALC256_FIXUP_ASUS_MIC is not needed anymore for ASUS Vivobook X705UD / N705UD laptops. Without it, everything works fine: - the internal microphone is detected and records actual sound, - plugging in a jack headset is detected and can record actual sound with it, - unplugging the jack headset makes the system go back to internal microphone and can record actual sound. Cc: stable@vger.kernel.org Cc: Kuan-Wei Chiu Cc: Chris Chiu Fixes: 3b4309546b48 ("ALSA: hda: Fix headset detection failure due to unstable sort") Tested-by: Adrien Vergé Signed-off-by: Adrien Vergé Link: https://patch.msgid.link/20250226135515.24219-1-adrienverge@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e5c80d4be535..c735f630ecb5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10657,7 +10657,6 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), - SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1a63, "ASUS UX3405MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2),