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[PATCH AUTOSEL 6.9 18/40] ASoC: cs35l56: Disconnect ASP1 TX sources when ASP1 DAI is hooked up
by Sasha Levin 09 Jul '24

09 Jul '24

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 8af49868e51ed1ba117b74728af12abe1eda82e5 ] If the ASP1 DAI is hooked up by the machine driver the ASP TX mixer sources should be initialized to disconnected. There aren't currently any available products using the ASP so this doesn't affect any existing systems. The cs35l56 does not have any fixed default for the mixer source registers. When the cs35l56 boots, its firmware patches these registers to setup a system-specific routing; this is so that Windows can use generic SDCA drivers instead of needing knowledge of chip-specific registers. The setup varies between end-products, which each have customized firmware, and so the default register state varies between end-products. It can also change if the firmware on an end-product is upgraded - for example if a change was needed to the routing for Windows use-cases. It must be emphasized that the settings applied by the firmware are not internal magic tuning; they are statically implementing use-case setup that on Linux would be done via ALSA controls. The driver is currently syncing the mixer controls with whatever initial state the firmware wrote to the registers, so that they report the actual audio routing. But if the ASP DAI is hooked up this can create a powered-up DAPM graph without anything intentionally setting up a path. This can lead to parts of the audio system powering up unexpectedly. For example when cs35l56 is connected to cs42l43 using a codec-codec link, this can create a complete DAPM graph which then powers-up cs42l43. But the cs42l43 can only be clocked from its SoundWire bus so this causes a bunch of errors in the kernel log where cs42l43 is unexpectedly powered-up without a clock. If the host is taking ownership of the ASP (either directly or as a codec-to-codec link) there is no need to keep the mixer settings that the firmware wrote. The driver has ALSA controls for setting these using standard Linux mechanisms. So if the machine driver hooks up the ASP the ASP mixers are initialized to "None" (no input). This prevents unintended DAPM-graph power-ups, and means the initial state of the mixers is always going to be None. Since the initial state of the mixers can vary from system to system and potentially between firmware upgrades, no use-case manager can currently assume that cs35l56 has a known initial state. The firmware could just as easily default them to "None" as to any input source. So defaulting them to "None" in the driver is not increasing the entropy of the system. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20240613132527.46537-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l56-shared.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index fd02b621da52c..d29878af2a80d 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -214,6 +214,10 @@ static const struct reg_sequence cs35l56_asp1_defaults[] = { REG_SEQ0(CS35L56_ASP1_FRAME_CONTROL5, 0x00020100), REG_SEQ0(CS35L56_ASP1_DATA_CONTROL1, 0x00000018), REG_SEQ0(CS35L56_ASP1_DATA_CONTROL5, 0x00000018), + REG_SEQ0(CS35L56_ASP1TX1_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX2_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX3_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX4_INPUT, 0x00000000), }; /* -- 2.43.0

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[PATCH AUTOSEL 6.9 14/40] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 4513b527ab970..ad8925b6481ca 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

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[PATCH AUTOSEL 6.9 13/40] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 1e760c3155213..2b1ed91a736c9 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1472,10 +1472,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1490,6 +1491,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1589,9 +1591,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

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[PATCH AUTOSEL 6.9 11/40] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index b41a1147f1c34..a64d1989e28a5 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

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[PATCH AUTOSEL 6.9 08/40] ASoC: Intel: avs: Fix route override
by Sasha Levin 09 Jul '24

09 Jul '24

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit fd660b1bd015e5aa9a558ee04088f2431010548d ] Instead of overriding existing memory strings that may be too short, just allocate needed memory and point the route at it. Reported-by: Jason Montleon <jmontleo(a)redhat.com> Link: https://github.com/thesofproject/avs-topology-xml/issues/22#issuecomment-21… 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/20240603102818.36165-3-amadeuszx.slawinski@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/topology.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 42b42903ae9de..691d16ce95a0f 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -1545,8 +1545,8 @@ static int avs_route_load(struct snd_soc_component *comp, int index, { struct snd_soc_acpi_mach *mach = dev_get_platdata(comp->card->dev); size_t len = SNDRV_CTL_ELEM_ID_NAME_MAXLEN; - char buf[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int ssp_port, tdm_slot; + char *buf; /* See parse_link_formatted_string() for dynamic naming when(s). */ if (!avs_mach_singular_ssp(mach)) @@ -1557,13 +1557,24 @@ static int avs_route_load(struct snd_soc_component *comp, int index, return 0; tdm_slot = avs_mach_ssp_tdm(mach, ssp_port); + buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL); + if (!buf) + return -ENOMEM; avs_ssp_sprint(buf, len, route->source, ssp_port, tdm_slot); - strscpy((char *)route->source, buf, len); + route->source = buf; + + buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL); + if (!buf) + return -ENOMEM; avs_ssp_sprint(buf, len, route->sink, ssp_port, tdm_slot); - strscpy((char *)route->sink, buf, len); + route->sink = buf; + if (route->control) { + buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL); + if (!buf) + return -ENOMEM; avs_ssp_sprint(buf, len, route->control, ssp_port, tdm_slot); - strscpy((char *)route->control, buf, len); + route->control = buf; } return 0; -- 2.43.0

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Mic not working on "ROG Strix Z790-A Gaming WIFI II" with ALC4080
by GitHub issues - edited 09 Jul '24

09 Jul '24

alsa-project/alsa-ucm-conf issue #431 was edited from luluonino: I ran into this issue https://github.com/alsa-project/alsa-ucm-conf/issues/394 while searching about exactly the same problem on "ROG Strix Z790-A Gaming WIFI II". The commit included in issue 394 fixed the speaker on my headphone plugged into the case front panel. But the mic still doesn't work. It is detected as "Microphone - USB Audio" but it doesn't pick up any sound. I'm running Ubuntu 24.04 and with alsa-ucm-conf-1.2.12 applied. Here is the alsa-info dump. Note after applying the fix in issue 394, the audio codec is still detected as generic USB audio but there is indeed sound afterwards. [alsa-info.txt](https://github.com/user-attachments/files/16136523/alsa-info… I've tried all potential solutions I could find online but none worked. Any help is greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/431 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/realtek: Enable Mute LED on HP 250 G7
by Nazar Bilinskyi 09 Jul '24

09 Jul '24

HP 250 G7 has a mute LED that can be made to work using quirk ALC269_FIXUP_HP_LINE1_MIC1_LED. Enable already existing quirk. Signed-off-by: Nazar Bilinskyi <nbilinskyi(a)gmail.com> --- 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 811e82474..57eb67f05 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10053,6 +10053,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x84a6, "HP 250 G7 Notebook PC", ALC269_FIXUP_HP_LINE1_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x84ae, "HP 15-db0403ng", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), -- 2.45.2

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Mic not working on "ROG Strix Z790-A Gaming WIFI II" with ALC4080
by GitHub issues - edited 09 Jul '24

09 Jul '24

alsa-project/alsa-ucm-conf issue #431 was edited from luluonino: I ran into this issue https://github.com/alsa-project/alsa-ucm-conf/issues/394 while searching about exactly the same problem on "ROG Strix Z790-A Gaming WIFI II". The commit included in issue 394 fixed the speaker on my microphone plugged into the case front panel. But the mic still doesn't work. It is detected as "Microphone - USB Audio" but it doesn't pick up any sound. I'm running Ubuntu 24.04 and with alsa-ucm-conf-1.2.12 applied. Here is the alsa-info dump. Note after applying the fix in issue 394, the audio codec is still detected as generic USB audio but there is indeed sound afterwards. [alsa-info.txt](https://github.com/user-attachments/files/16136523/alsa-info… I've tried all potential solutions I could find online but none worked. Any help is greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/431 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Mic not working on "ROG Strix Z790-A Gaming WIFI II" with ALC4080
by GitHub issues - opened 09 Jul '24

09 Jul '24

alsa-project/alsa-ucm-conf issue #431 was opened from luluonino: I ran into this issue https://github.com/alsa-project/alsa-ucm-conf/issues/394 while searching about exactly the same problem on "ROG Strix Z790-A Gaming WIFI II". The commit included in issue 394 fixed the speaker on my microphone plugged into the case front panel. But the mic still doesn't work. It is detected as "Microphone - USB Audio" but it doesn't pick up any sound. I'm running Ubuntu 24.04. Here is the alsa-info dump. Note after applying the fix in issue 394, the audio codec is still detected as generic USB audio but there is indeed sound afterwards. [alsa-info.txt](https://github.com/user-attachments/files/16136523/alsa-info… I've tried all potential solutions I could find online but none worked. Any help is greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/431 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Re: (subset) [PATCH v6 0/7] ASoC: fsl-asoc-card: add S/PDIF controller support
by Mark Brown 08 Jul '24

08 Jul '24

On Thu, 27 Jun 2024 10:30:57 +0200, Elinor Montmasson wrote: > This is the v6 of the series of patches aiming to make the machine > driver `fsl-asoc-card` compatible with S/PDIF controllers on imx boards. > The main goal is to allow the use of S/PDIF controllers with ASRC > modules. > > The `imx-spdif` machine driver already has specific support for S/PDIF > controllers but doesn't support using an ASRC with it. However, the > `fsl-asoc-card` machine driver has the necessary code to create a sound > card which can use an ASRC module. > It is then possible to extend the support for S/PDIF audio cards by > merging the `imx-spdif` driver into `fsl-asoc-card`. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/7] ASoC: fsl-asoc-card: add support for dai links with multiple codecs commit: a613b63db233b6e7d46ec72f009c1cbb7db8be66 [2/7] ASoC: fsl-asoc-card: add second dai link component for codecs commit: c68fa0d9b0f8cc7c3ae7d29c02adbc97622a73f5 [3/7] ASoC: fsl-asoc-card: add compatibility to use 2 codecs in dai-links commit: fcc6ace84f1f6ce2211af25c3c8fb30a0fb2bb2c [4/7] ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c commit: 6d174cc4f22461ad3fe383570527e86bf1948a2e [5/7] ASoC: dt-bindings: update fsl-asoc-card bindings after imx-spdif merge commit: 4359caadd17fbde64d656c10bd6f2dc91b675a11 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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