October 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: SOF: Intel: hda-stream: limit PROCEN workaround
by Peter Ujfalusi 01 Oct '21

01 Oct '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The work-around enabled in hda-stream.c is only required on earlier versions of SOCs/PCH (Skylake, KabyLake, ApolloLake, GeminiLake). Before setting the format on the host DMA, it is required to couple the host and link DMA - which as a consequence shall use the same format. This patch introduces a quirk field in the platform descriptor and makes the work-around conditional. Newer platforms have no limitations on the use of host and link DMA, which can use different formats. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/hda-stream.c | 18 ++++++++++++------ sound/soc/sof/intel/shim.h | 4 ++++ 3 files changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index e6a1f6532547..16b7d8b1adff 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -143,5 +143,6 @@ const struct sof_intel_dsp_desc apl_chip_info = { .rom_init_timeout = 150, .ssp_count = APL_SSP_COUNT, .ssp_base_offset = APL_SSP_BASE_OFFSET, + .quirks = SOF_INTEL_PROCEN_FMT_QUIRK, }; EXPORT_SYMBOL_NS(apl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 63c367478f1c..47a465b06da5 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -405,6 +405,7 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, struct snd_pcm_hw_params *params) { + const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); struct hdac_bus *bus = sof_to_bus(sdev); struct hdac_stream *hstream = &stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); @@ -538,6 +539,7 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, /* * Recommended hardware programming sequence for HDAudio DMA format + * on earlier platforms - this is not needed on newer platforms * * 1. Put DMA into coupled mode by clearing PPCTL.PROCEN bit * for corresponding stream index before the time of writing @@ -547,9 +549,11 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, * enable decoupled mode */ - /* couple host and link DMA, disable DSP features */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - mask, 0); + if (chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK) { + /* couple host and link DMA, disable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, 0); + } /* program stream format */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, @@ -557,9 +561,11 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, SOF_HDA_ADSP_REG_CL_SD_FORMAT, 0xffff, hstream->format_val); - /* decouple host and link DMA, enable DSP features */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - mask, mask); + if (chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK) { + /* decouple host and link DMA, enable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, mask); + } /* program last valid index */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index e9f7d4d7fcce..8e2a0f4c0f66 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -151,6 +151,9 @@ #define PCI_PMCS 0x84 #define PCI_PMCS_PS_MASK 0x3 +/* Intel quirks */ +#define SOF_INTEL_PROCEN_FMT_QUIRK BIT(0) + /* DSP hardware descriptor */ struct sof_intel_dsp_desc { int cores_num; @@ -166,6 +169,7 @@ struct sof_intel_dsp_desc { int ssp_base_offset; /* base address of the SSPs */ u32 sdw_shim_base; u32 sdw_alh_base; + u32 quirks; bool (*check_sdw_irq)(struct snd_sof_dev *sdev); }; -- 2.33.0

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[PATCH][next] ALSA: usb-audio: Fix sum of uninitialized variable sample_accum
by Colin King 01 Oct '21

01 Oct '21

From: Colin Ian King <colin.king(a)canonical.com> Variable sample_accum is not being intialized and then has ep->sample_rem added to it, leading to a bogus value. One solution is to initialize it to zero at declaration time, but it is probably best to just assign it to ep->sample_rem on first use. Addresses-Coveriry: ("Uninitialized scalar variable") Fixes: f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 42c0d2db8ba8..c6a33732db3f 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -182,7 +182,7 @@ static int next_packet_size(struct snd_usb_endpoint *ep, unsigned int avail) if (ep->fill_max) return ep->maxframesize; - sample_accum += ep->sample_rem; + sample_accum = ep->sample_rem; if (sample_accum >= ep->pps) { sample_accum -= ep->pps; ret = ep->packsize[1]; -- 2.32.0

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[PATCH] ALSA: usb-audio: Fix packet size calculation regression
by Takashi Iwai 01 Oct '21

01 Oct '21

The commit d215f63d49da ("ALSA: usb-audio: Check available frames for the next packet size") introduced the available frame size check, but the conversion forgot to initialize the temporary variable properly, and it resulted in a bogus calculation. This patch fixes it. Fixes: d215f63d49da ("ALSA: usb-audio: Check available frames for the next packet size") Reported-by: Colin Ian King <colin.king(a)canonical.com> Link: https://lore.kernel.org/r/20211001104417.14291-1-colin.king@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 42c0d2db8ba8..743b8287cfcd 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -182,7 +182,7 @@ static int next_packet_size(struct snd_usb_endpoint *ep, unsigned int avail) if (ep->fill_max) return ep->maxframesize; - sample_accum += ep->sample_rem; + sample_accum = ep->sample_accum + ep->sample_rem; if (sample_accum >= ep->pps) { sample_accum -= ep->pps; ret = ep->packsize[1]; -- 2.26.2

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ALSA kernel projects - for academic purposes
by Muni Sekhar 01 Oct '21

01 Oct '21

Hi all, What small projects would you suggest to a novice with the ALSA kernel. The aim is to develop a familiarity with the ALSA kernel source code, and also to submit it for academic purposes. -- Thanks, Sekhar

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[PATCH v2 0/2] ASoC: mediatek: Add support for MT8195 sound card with rt1011 and rt5682
by Trevor Wu 01 Oct '21

01 Oct '21

This series of patches adds support for mt8195 board with mt6359, rt1011 and rt5682. Patches are based on broonie tree "for-next" branch. Changes since v1: - refined code based on Pierre-Louis's suggestions - added missing of_node_put - fixed wrong fixup assignment on HDMITX Trevor Wu (2): ASoC: mediatek: mt8195: add machine driver with mt6359, rt1011 and rt5682 dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1011-rt5682 document .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 47 + sound/soc/mediatek/Kconfig | 15 + sound/soc/mediatek/mt8195/Makefile | 1 + .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 1155 +++++++++++++++++ 4 files changed, 1218 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1011-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c -- 2.18.0

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[PATCH] ALSA: hda - Enable headphone mic on Dell Latitude laptops with ALC3254
by Chris Chiu 01 Oct '21

01 Oct '21

The headphone mic is not working on Dell Latitude laptops with ALC3254. The codec vendor id is 0x10ec0295 and share the same pincfg as defined in ALC295_STANDARD_PINS. So the ALC269_FIXUP_DELL1_MIC_NO_PRESENCE will be applied per alc269_pin_fixup_tbl[] but actually the headphone mic is using NID 0x1b instead of 0x1a. The ALC269_FIXUP_DELL4_MIC_NO_PRESENCE need to be applied instead. Use ALC269_FIXUP_DELL4_MIC_NO_PRESENCE for particular models before a generic fixup comes out. Signed-off-by: Chris Chiu <chris.chiu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4407f7da57c4..70778aa137cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8452,6 +8452,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.20.1

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[PATCH] ALSA: usb-audio: disable implicit feedback sync for Behringer UFX1204 and UFX1604
by Geraldo Nascimento 01 Oct '21

01 Oct '21

Hi Takashi, Hopefully you will be able to apply this cleanly to branch master of your sound.git. Thanks! Geraldo Nascimento --- Behringer UFX1204 and UFX1604 have Synchronous endpoints to which current ALSA code applies implicit feedback sync as if they were Asynchronous endpoints. This breaks UAC compliance and is unneeded. The commit 5e35dc0338d85ccebacf3f77eca1e5dea73155e8 and subsequent 1a15718b41df026cffd0e42cfdc38a1384ce19f9 were meant to clear up noise. Unfortunately, noise persisted for those using higher sample rates and this was only solved by commit d2e8f641257d0d3af6e45d6ac2d6f9d56b8ea964 Since there are no more reports of noise, let's get rid of the implicit-fb quirks breaking UAC compliance. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 23767a14d126..70319c822c10 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -54,8 +54,6 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Fixed EP */ /* FIXME: check the availability of generic matching */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0001, 0x81, 1), /* Behringer UFX1604 */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0002, 0x81, 1), /* Behringer UFX1204 */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8010, 0x81, 2), /* Fractal Audio Axe-Fx III */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0001, 0x81, 2), /* Solid State Logic SSL2 */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0002, 0x81, 2), /* Solid State Logic SSL2+ */

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[PATCH v2] soundwire: debugfs: use controller id and link_id for debugfs
by Srinivas Kandagatla 01 Oct '21

01 Oct '21

link_id can be zero and if we have multiple controller instances in a system like Qualcomm debugfs will end-up with duplicate namespace resulting in incorrect debugfs entries. Using bus-id and link-id combination should give a unique debugfs directory entry and should fix below warning too. "debugfs: Directory 'master-0' with parent 'soundwire' already present!" Fixes: bf03473d5bcc ("soundwire: add debugfs support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: Added Link ID along with bus id. drivers/soundwire/debugfs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/debugfs.c b/drivers/soundwire/debugfs.c index b6cad0d59b7b..49900cd207bc 100644 --- a/drivers/soundwire/debugfs.c +++ b/drivers/soundwire/debugfs.c @@ -19,7 +19,7 @@ void sdw_bus_debugfs_init(struct sdw_bus *bus) return; /* create the debugfs master-N */ - snprintf(name, sizeof(name), "master-%d", bus->link_id); + snprintf(name, sizeof(name), "master-%d-%d", bus->id, bus->link_id); bus->debugfs = debugfs_create_dir(name, sdw_debugfs_root); } -- 2.21.0

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