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[PATCH] ASoC: dt-bindings: rt1015p: fix syntax error in dts-binding document
by Jack Yu 10 Aug '21

10 Aug '21

Fix syntax error in dts-binding document. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 8fcb5f79a1b5..f31d3c4d0192 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -16,8 +16,8 @@ description: | properties: compatible: oneOf: - const: realtek,rt1015p - const: realtek,rt1019p + - const: realtek,rt1015p + - const: realtek,rt1019p sdb-gpios: description: -- 2.31.1

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[PATCH 1/2] ALSA: hda: Release controller display power during shutdown/reboot
by Imre Deak 10 Aug '21

10 Aug '21

Make sure the HDA driver's display power reference is released during shutdown/reboot. During the shutdown/reboot sequence the pci device core calls the pm_runtime_resume handler for all devices before calling the driver's shutdown callback and so the HDA driver's runtime resume callback will acquire a display power reference (on HSW/BDW). This triggers a power reference held WARN on HSW/BDW in the i915 driver's subsequent shutdown handler, which expects all display power references to be released by that time. Since the HDA controller is stopped in the shutdown handler in any case, let's follow here the same sequence as the one during runtime suspend. This will also reset the HDA link and drop the display power reference, getting rid of the above WARN. Tested on HSW. v2: - Fix the build for CONFIG_PM=n (Takashi) - s/__azx_runtime_suspend/azx_shutdown_chip/ Closes: https://gitlab.freedesktop.org/drm/intel/-/issues/3618 References: https://lore.kernel.org/lkml/cea1f9a-52e0-b83-593d-52997fe1aaf6@er-systems.… Reported-and-tested-by: Thomas Voegtle <tv(a)lio96.de> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Imre Deak <imre.deak(a)intel.com> --- sound/pci/hda/hda_intel.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7f8f11536a3dc..f5ab0b682adcc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -883,6 +883,14 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } +static void azx_shutdown_chip(struct azx *chip) +{ + azx_stop_chip(chip); + azx_enter_link_reset(chip); + azx_clear_irq_pending(chip); + display_power(chip, false); +} + #ifdef CONFIG_PM static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); @@ -942,14 +950,6 @@ static bool azx_is_pm_ready(struct snd_card *card) return true; } -static void __azx_runtime_suspend(struct azx *chip) -{ - azx_stop_chip(chip); - azx_enter_link_reset(chip); - azx_clear_irq_pending(chip); - display_power(chip, false); -} - static void __azx_runtime_resume(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -1028,7 +1028,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); - __azx_runtime_suspend(chip); + azx_shutdown_chip(chip); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1107,7 +1107,7 @@ static int azx_runtime_suspend(struct device *dev) /* enable controller wake up event */ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | STATESTS_INT_MASK); - __azx_runtime_suspend(chip); + azx_shutdown_chip(chip); trace_azx_runtime_suspend(chip); return 0; } @@ -2383,7 +2383,7 @@ static void azx_shutdown(struct pci_dev *pci) return; chip = card->private_data; if (chip && chip->running) - azx_stop_chip(chip); + azx_shutdown_chip(chip); } /* PCI IDs */ -- 2.27.0

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[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 650 G8 Notebook PC
by Jeremy Szu 10 Aug '21

10 Aug '21

The HP ProBook 650 G8 Notebook PC is using ALC236 codec which is using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.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 4043a2362f27..a065260d0d20 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8431,6 +8431,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), + SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.25.1

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[PATCH RFC 0/5] ALSA: Support for non-coherent and non-contiguous page allocation
by Takashi Iwai 10 Aug '21

10 Aug '21

Hi, this is an experimental patch set to add the support for PCM buffers with non-coherent and non-contiguous pages, typically useful for non-x86 architectures. The first patch improves the SG-buffer handling, then add a new PCM info flag that disables the control and status mmap, and implements two new buffer types, SNDRV_DMA_TYPE_NONCONTIG and SNDRV_DMA_TYPE_NONCOHERENT. The former is the SG-buffer and the latter is the continuous page allocation, corresponding to SNDRV_DMA_TYPE_DEV_SG and SNDRV_DMA_TYPE_DEV on x86. Unlike other page types, those are directional (that need the DMA direction at allocation time) and require the explicit sync of buffers around the data transfer (flushing, invalidating). The sync is implemented inside ALSA PCM core and automatically applied at updating the applptr and hwsync via SYNC_PTR ioctl, which should be issued during mmap operation, so it should work transparently as long as applications are running with alsa-lib. For tinyALSA, we might need to revisit the implementation. This is currently an RFC -- more exactly, CALL FOR TESTERS, as I have no Arm machine with the sound device for now. The needed change for an existing driver is simple: * Replace the buffer type from SNDRV_DMA_TYPE_DEV to SNDRV_DMA_TYPE_NONCOHERENT (or from SNDRV_DMA_TYPE_DEV_SG to SNDRV_DMA_TYPE_NONCONTIG), which is set up typically in snd_pcm_set_managed_buffer*() call. * Add SNDRV_PCM_INFO_EXPLICIT_SYNC flag to the PCM runtime hardware.info field It'd be greatly appreciated if anyone can test and try the changes, and help debugging. Again, the code is faily untested. The latest code is found in topic/memalloc-noncontig branch of my sound git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git The patch set is based on for-next branch, i.e. changes for 5.15 are included. thanks, Takashi === Takashi Iwai (5): ALSA: memalloc: Count continuous pages in vmalloc buffer handler ALSA: pcm: Add SNDRV_PCM_INFO_EXPLICIT_SYNC flag ALSA: memalloc: Assign ops field to snd_dma_buffer ALSA: memalloc: Support for non-contiguous page allocation ALSA: memalloc: Support for non-coherent page allocation include/sound/memalloc.h | 48 +++++++- include/uapi/sound/asound.h | 1 + sound/core/memalloc.c | 223 ++++++++++++++++++++++++++++-------- sound/core/memalloc_local.h | 1 + sound/core/pcm_lib.c | 9 ++ sound/core/pcm_memory.c | 13 ++- sound/core/pcm_native.c | 22 ++++ 7 files changed, 263 insertions(+), 54 deletions(-) -- 2.26.2

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[PATCH] ASoC: Intel: Fix platform ID matching
by Curtis Malainey 09 Aug '21

09 Aug '21

Sparse warnings triggered truncating the IDs of some platform device tables. Unfortunately some of the IDs in the match tables were missed which breaks audio. The KBL change has been verified to fix audio, the CML change was not tested as it was found through grepping the broken changes and found to match the same situation in anticipation that it should also be fixed. Fixes: 94efd726b947 ("ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters") Fixes: 24e46fb811e9 ("ASoC: Intel: bxt_da7219_max98357a: shrink platform_id below 20 characters") Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Tested-by: Matt Davis <mattedavis(a)chromium.org> --- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 42ef51c3fb4f..b591c6fd13fd 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -75,7 +75,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { }, { .id = "DLGS7219", - .drv_name = "cml_da7219_max98357a", + .drv_name = "cml_da7219_mx98357a", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &max98390_spk_codecs, .sof_fw_filename = "sof-cml.ri", diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index ba5ff468c265..741bf2f9e081 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -87,7 +87,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_max98357a", + .drv_name = "kbl_da7219_mx98357a", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98357_codecs, -- 2.32.0.605.g8dce9f2422-goog

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[PATCH 1/2] ASoC: rt1015p: add new acpi id and comapatible id
by Jack Yu 09 Aug '21

09 Aug '21

Add new acpi id and compatible id for rt1015p. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015p.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 40f2063aefbe..415cfb3b2f0d 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -127,6 +127,7 @@ static int rt1015p_platform_probe(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id rt1015p_device_id[] = { { .compatible = "realtek,rt1015p" }, + { .compatible = "realtek,rt1019p" }, {} }; MODULE_DEVICE_TABLE(of, rt1015p_device_id); @@ -135,6 +136,7 @@ MODULE_DEVICE_TABLE(of, rt1015p_device_id); #ifdef CONFIG_ACPI static const struct acpi_device_id rt1015p_acpi_match[] = { { "RTL1015", 0}, + { "RTL1019", 0}, { }, }; MODULE_DEVICE_TABLE(acpi, rt1015p_acpi_match); -- 2.31.1

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[PATCH v2] ASoC: max98390: Add support change dsm param name
by Steve Lee 09 Aug '21

09 Aug '21

In case of using different type of speaker, support using different dsm parameter bin file for each amp connected. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 26 ++++++++++++++++++++------ sound/soc/codecs/max98390.h | 1 + 2 files changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 94773ccee9d5..b392567c2b3e 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -765,17 +765,26 @@ static int max98390_dsm_init(struct snd_soc_component *component) vendor = dmi_get_system_info(DMI_SYS_VENDOR); product = dmi_get_system_info(DMI_PRODUCT_NAME); - if (vendor && product) { - snprintf(filename, sizeof(filename), "dsm_param_%s_%s.bin", - vendor, product); + if (!strcmp(max98390->dsm_param_name, "default")) { + if (vendor && product) { + snprintf(filename, sizeof(filename), + "dsm_param_%s_%s.bin", vendor, product); + } else { + sprintf(filename, "dsm_param.bin"); + } } else { - sprintf(filename, "dsm_param.bin"); + snprintf(filename, sizeof(filename), "%s", + max98390->dsm_param_name); } ret = request_firmware(&fw, filename, component->dev); if (ret) { ret = request_firmware(&fw, "dsm_param.bin", component->dev); - if (ret) - goto err; + if (ret) { + ret = request_firmware(&fw, "dsmparam.bin", + component->dev); + if (ret) + goto err; + } } dev_dbg(component->dev, @@ -1047,6 +1056,11 @@ static int max98390_i2c_probe(struct i2c_client *i2c, __func__, max98390->ref_rdc_value, max98390->ambient_temp_value); + ret = device_property_read_string(&i2c->dev, "maxim,dsm_param_name", + &max98390->dsm_param_name); + if (ret) + max98390->dsm_param_name = "default"; + /* voltage/current slot configuration */ max98390_slot_config(i2c, max98390); diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index e31516717d3b..c250740f73a2 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -662,5 +662,6 @@ struct max98390_priv { unsigned int i_l_slot; unsigned int ref_rdc_value; unsigned int ambient_temp_value; + const char *dsm_param_name; }; #endif -- 2.17.1

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[PATCH 2/2] ASoC: dt-bindings: rt1015p: add new compatible id
by Jack Yu 09 Aug '21

09 Aug '21

Add new compatible ID for rt1015p in dt-bindings document. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 644b68edf3e1..8fcb5f79a1b5 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -15,7 +15,9 @@ description: | properties: compatible: - const: realtek,rt1015p + oneOf: + const: realtek,rt1015p + const: realtek,rt1019p sdb-gpios: description: -- 2.31.1

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[PATCH] ASoC: rt1015p: add new acpi id and comapatible id
by Jack Yu 09 Aug '21

09 Aug '21

Add new acpi id and compatible id for rt1015p. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 4 +++- sound/soc/codecs/rt1015p.c | 2 ++ 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 644b68edf3e1..b898dfa4eb83 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -15,7 +15,9 @@ description: | properties: compatible: - const: realtek,rt1015p + oneOf: + - const: realtek,rt1015p + - const: realtek,rt1019p sdb-gpios: description: diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 39f4c4ed2ab8..085a7b3c7bb4 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -126,6 +126,7 @@ static int rt1015p_platform_probe(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id rt1015p_device_id[] = { { .compatible = "realtek,rt1015p" }, + { .compatible = "realtek,rt1019p" }, {} }; MODULE_DEVICE_TABLE(of, rt1015p_device_id); @@ -134,6 +135,7 @@ MODULE_DEVICE_TABLE(of, rt1015p_device_id); #ifdef CONFIG_ACPI static const struct acpi_device_id rt1015p_acpi_match[] = { { "RTL1015", 0}, + { "RTL1019", 0}, { }, }; MODULE_DEVICE_TABLE(acpi, rt1015p_acpi_match); -- 2.29.0

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[PATCH] soundwire: intel: trap TRIGGER_SUSPEND in .trigger callback
by Bard Liao 09 Aug '21

09 Aug '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> This patch provides both a simplification of the suspend flows and a better balanced operation during suspend/resume transition. The exiting code relies on a convoluted way of dealing with suspend signals. Since there is no .suspend DAI callback, we used the component .suspend and marked all the component DAI dmas as 'suspended'. The information was used in the .prepare stage to differentiate resume operations from xrun handling, and only reinitialize SHIM registers and DMA in the former case. While this solution has been working reliably for about 2 years, there is a much better solution consisting in trapping the TRIGGER_SUSPEND in the .trigger DAI ops. The DMA is still marked in the same way for the .prepare op to run, but in addition the callbacks sent to DSP firmware are now balanced. Normal operation: hw_params -> intel_params_stream hw_free -> intel_free_stream suspend -> intel_free_stream prepare -> intel_params_stream This balanced operation was not required with existing SOF firmware relying on static pipelines instantiated at every boot. With the on-going transition to dynamic pipelines, it's however a requirement to keep the use count for the DAI widget balanced across all transitions. Co-developed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 53 +++++++++++++++++++++------------------ 1 file changed, 29 insertions(+), 24 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index fb9c23e13206..a0178779a5ba 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1005,29 +1005,6 @@ static void intel_shutdown(struct snd_pcm_substream *substream, pm_runtime_put_autosuspend(cdns->dev); } -static int intel_component_dais_suspend(struct snd_soc_component *component) -{ - struct sdw_cdns_dma_data *dma; - struct snd_soc_dai *dai; - - for_each_component_dais(component, dai) { - /* - * we don't have a .suspend dai_ops, and we don't have access - * to the substream, so let's mark both capture and playback - * DMA contexts as suspended - */ - dma = dai->playback_dma_data; - if (dma) - dma->suspended = true; - - dma = dai->capture_dma_data; - if (dma) - dma->suspended = true; - } - - return 0; -} - static int intel_pcm_set_sdw_stream(struct snd_soc_dai *dai, void *stream, int direction) { @@ -1056,11 +1033,39 @@ static void *intel_get_sdw_stream(struct snd_soc_dai *dai, return dma->stream; } +static int intel_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_intel *sdw = cdns_to_intel(cdns); + struct sdw_cdns_dma_data *dma; + + /* + * The .prepare callback is used to deal with xruns and resume operations. In the case + * of xruns, the DMAs and SHIM registers cannot be touched, but for resume operations the + * DMAs and SHIM registers need to be initialized. + * the .trigger callback is used to track the suspend case only. + */ + if (cmd != SNDRV_PCM_TRIGGER_SUSPEND) + return 0; + + dma = snd_soc_dai_get_dma_data(dai, substream); + if (!dma) { + dev_err(dai->dev, "failed to get dma data in %s\n", + __func__); + return -EIO; + } + + dma->suspended = true; + + return intel_free_stream(sdw, substream, dai, sdw->instance); +} + static const struct snd_soc_dai_ops intel_pcm_dai_ops = { .startup = intel_startup, .hw_params = intel_hw_params, .prepare = intel_prepare, .hw_free = intel_hw_free, + .trigger = intel_trigger, .shutdown = intel_shutdown, .set_sdw_stream = intel_pcm_set_sdw_stream, .get_sdw_stream = intel_get_sdw_stream, @@ -1071,6 +1076,7 @@ static const struct snd_soc_dai_ops intel_pdm_dai_ops = { .hw_params = intel_hw_params, .prepare = intel_prepare, .hw_free = intel_hw_free, + .trigger = intel_trigger, .shutdown = intel_shutdown, .set_sdw_stream = intel_pdm_set_sdw_stream, .get_sdw_stream = intel_get_sdw_stream, @@ -1078,7 +1084,6 @@ static const struct snd_soc_dai_ops intel_pdm_dai_ops = { static const struct snd_soc_component_driver dai_component = { .name = "soundwire", - .suspend = intel_component_dais_suspend }; static int intel_create_dai(struct sdw_cdns *cdns, -- 2.17.1

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