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[PATCH v1] slimbus: Use dev_err_probe instead of dev_err
by 王明-软件底层技术部 19 Jul '23

19 Jul '23

It is possible that dma_request_chan will return EPROBE_DEFER, which means that dev is not ready yet. In this case, dev_err(dev), there will be no output. This patch fixes the bug. Signed-off-by: Wang Ming <machel(a)vivo.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 77aa6d26476c..d5c2a732f63f 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -679,8 +679,8 @@ static int qcom_slim_ngd_init_rx_msgq(struct qcom_slim_ngd_ctrl *ctrl) ctrl->dma_rx_channel = dma_request_chan(dev, "rx"); if (IS_ERR(ctrl->dma_rx_channel)) { - dev_err(dev, "Failed to request RX dma channel"); - ret = PTR_ERR(ctrl->dma_rx_channel); + ret = dev_err_probe(dev, PTR_ERR(ctrl->dma_rx_channel), + "Failed to request RX dma channel"); ctrl->dma_rx_channel = NULL; return ret; } @@ -717,8 +717,8 @@ static int qcom_slim_ngd_init_tx_msgq(struct qcom_slim_ngd_ctrl *ctrl) ctrl->dma_tx_channel = dma_request_chan(dev, "tx"); if (IS_ERR(ctrl->dma_tx_channel)) { - dev_err(dev, "Failed to request TX dma channel"); - ret = PTR_ERR(ctrl->dma_tx_channel); + ret = dev_err_probe(dev, PTR_ERR(ctrl->dma_tx_channel), + "Failed to request TX dma channel"); ctrl->dma_tx_channel = NULL; return ret; } -- 2.25.1

1 0

Correct Home For Documentation of ALSA's PCM State Machine
by Chas Leichner 19 Jul '23

19 Jul '23

Hi all, Several years ago, I went through the kernel and documented how ALSA API calls influence the PCM state. (https://source.chromium.org/chromium/chromium/src/+/main:chromecast/media/c…) One of my coworkers noted that this was still a useful document for their work, so I thought it might be worthwhile to put it somewhere a bit more discoverable than deep in the Chromium source tree, but I don't see an obvious spot from poking around the ALSA wiki. Do you have a suggested location for it? Thank you, Chas

1 0

[PATCH 00/48] ASoC: cirrus: Use the maple tree register cache
by Mark Brown 19 Jul '23

19 Jul '23

The maple tree register register cache is now able to generate multi register writes which was the last big feature of the rbtree cache that it didn't support so let's update drivers to use the cache with the more modern data structure. --- Mark Brown (48): ASoC: cs35l36: Update to use maple tree register cache ASoC: cs35l41: Update to use maple tree register cache ASoC: cs35l45: Update to use maple tree register cache ASoC: cs35l56: Update to use maple tree register cache ASoC: cs42l51: Update to use maple tree register cache ASoC: cs4265: Update to use maple tree register cache ASoC: cs4270: Update to use maple tree register cache ASoC: cs42l52: Update to use maple tree register cache ASoC: cs42l56: Update to use maple tree register cache ASoC: cs42xx8: Update to use maple tree register cache ASoC: cs4349: Update to use maple tree register cache ASoC: wm2200: Update to use maple tree register cache ASoC: wm5100: Update to use maple tree register cache ASoC: wm9081: Update to use maple tree register cache ASoC: wm9090: Update to use maple tree register cache ASoC: wm8510: Update to use maple tree register cache ASoC: wm8523: Update to use maple tree register cache ASoC: wm8580: Update to use maple tree register cache ASoC: wm8711: Update to use maple tree register cache ASoC: wm8728: Update to use maple tree register cache ASoC: wm8731: Update to use maple tree register cache ASoC: wm8737: Update to use maple tree register cache ASoC: wm8741: Update to use maple tree register cache ASoC: wm8750: Update to use maple tree register cache ASoC: wm8753: Update to use maple tree register cache ASoC: wm8770: Update to use maple tree register cache ASoC: wm8776: Update to use maple tree register cache ASoC: wm8804: Update to use maple tree register cache ASoC: wm8900: Update to use maple tree register cache ASoC: wm8903: Update to use maple tree register cache ASoC: wm8904: Update to use maple tree register cache ASoC: wm8960: Update to use maple tree register cache ASoC: wm8961: Update to use maple tree register cache ASoC: wm8962: Update to use maple tree register cache ASoC: wm8991: Update to use maple tree register cache ASoC: wm8993: Update to use maple tree register cache ASoC: wm8995: Update to use maple tree register cache ASoC: wm8996: Update to use maple tree register cache ASoC: wm8940: Update to use maple tree register cache ASoC: wm8955: Update to use maple tree register cache ASoC: wm8971: Update to use maple tree register cache ASoC: wm8978: Update to use maple tree register cache ASoC: wm8983: Update to use maple tree register cache ASoC: wm8985: Update to use maple tree register cache ASoC: wm8988: Update to use maple tree register cache ASoC: wm9705: Update to use maple tree register cache ASoC: wm9712: Update to use maple tree register cache ASoC: wm9713: Update to use maple tree register cache sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41-lib.c | 4 ++-- sound/soc/codecs/cs35l45-tables.c | 4 ++-- sound/soc/codecs/cs35l56-shared.c | 6 +++--- sound/soc/codecs/cs4265.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/cs42xx8.c | 2 +- sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- 48 files changed, 52 insertions(+), 52 deletions(-) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230629-asoc-cirrus-maple-d3704cc6579f Best regards, -- Mark Brown <broonie(a)kernel.org>

2 51

[PATCH] ASoC: cs35l56: Patch soft registers to defaults
by Richard Fitzgerald 19 Jul '23

19 Jul '23

The soft (firmware) registers for volume/mute/posture are not reset by a chip soft-reset, so use a regmap patch to set them to defaults. cs35l56_reread_firmware_registers() has been removed. Its intent was to use whatever the firmware set as a default. But the driver now patches the defaults to the registers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- include/sound/cs35l56.h | 2 +- sound/soc/codecs/cs35l56-shared.c | 38 +++++++++++++------------------ sound/soc/codecs/cs35l56.c | 5 ++-- 3 files changed, 20 insertions(+), 25 deletions(-) diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index 1f9713d7ca76..ec672daa36cf 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -260,7 +260,7 @@ extern const struct cs_dsp_region cs35l56_dsp1_regions[CS35L56_NUM_DSP_REGIONS]; extern const char * const cs35l56_tx_input_texts[CS35L56_NUM_INPUT_SRC]; extern const unsigned int cs35l56_tx_input_values[CS35L56_NUM_INPUT_SRC]; -void cs35l56_reread_firmware_registers(struct device *dev, struct regmap *regmap); +int cs35l56_set_patch(struct regmap *regmap); int cs35l56_get_bclk_freq_id(unsigned int freq); void cs35l56_fill_supply_names(struct regulator_bulk_data *data); diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 60da8c75b7b9..7126f06b8047 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -11,6 +11,19 @@ #include "cs35l56.h" +static const struct reg_sequence cs35l56_patch[] = { + /* These are not reset by a soft-reset, so patch to defaults. */ + { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, + { CS35L56_MAIN_RENDER_USER_VOLUME, 0x00000000 }, + { CS35L56_MAIN_POSTURE_NUMBER, 0x00000000 }, +}; + +int cs35l56_set_patch(struct regmap *regmap) +{ + return regmap_register_patch(regmap, cs35l56_patch, ARRAY_SIZE(cs35l56_patch)); +} +EXPORT_SYMBOL_NS_GPL(cs35l56_set_patch, SND_SOC_CS35L56_SHARED); + static const struct reg_default cs35l56_reg_defaults[] = { { CS35L56_ASP1_ENABLES1, 0x00000000 }, { CS35L56_ASP1_CONTROL1, 0x00000028 }, @@ -35,9 +48,9 @@ static const struct reg_default cs35l56_reg_defaults[] = { { CS35L56_IRQ1_MASK_8, 0xfc000fff }, { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, { CS35L56_IRQ1_MASK_20, 0x15c00000 }, - /* CS35L56_MAIN_RENDER_USER_MUTE - soft register, no default */ - /* CS35L56_MAIN_RENDER_USER_VOLUME - soft register, no default */ - /* CS35L56_MAIN_POSTURE_NUMBER - soft register, no default */ + { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, + { CS35L56_MAIN_RENDER_USER_VOLUME, 0x00000000 }, + { CS35L56_MAIN_POSTURE_NUMBER, 0x00000000 }, }; static bool cs35l56_is_dsp_memory(unsigned int reg) @@ -181,25 +194,6 @@ static bool cs35l56_volatile_reg(struct device *dev, unsigned int reg) } } -static const u32 cs35l56_firmware_registers[] = { - CS35L56_MAIN_RENDER_USER_MUTE, - CS35L56_MAIN_RENDER_USER_VOLUME, - CS35L56_MAIN_POSTURE_NUMBER, -}; - -void cs35l56_reread_firmware_registers(struct device *dev, struct regmap *regmap) -{ - int i; - unsigned int val; - - for (i = 0; i < ARRAY_SIZE(cs35l56_firmware_registers); i++) { - regmap_read(regmap, cs35l56_firmware_registers[i], &val); - dev_dbg(dev, "%s: %d: %#x: %#x\n", __func__, - i, cs35l56_firmware_registers[i], val); - } -} -EXPORT_SYMBOL_NS_GPL(cs35l56_reread_firmware_registers, SND_SOC_CS35L56_SHARED); - const struct cs_dsp_region cs35l56_dsp1_regions[] = { { .type = WMFW_HALO_PM_PACKED, .base = CS35L56_DSP1_PMEM_0 }, { .type = WMFW_HALO_XM_PACKED, .base = CS35L56_DSP1_XMEM_PACKED_0 }, diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index c03f9d3c9a13..e046fdd26b74 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1572,8 +1572,9 @@ int cs35l56_init(struct cs35l56_private *cs35l56) if (ret) return ret; - /* Populate soft registers in the regmap cache */ - cs35l56_reread_firmware_registers(cs35l56->dev, cs35l56->regmap); + ret = cs35l56_set_patch(cs35l56->regmap); + if (ret) + return ret; /* Registers could be dirty after soft reset or SoundWire enumeration */ regcache_sync(cs35l56->regmap); -- 2.30.2

2 1

[PATCH] ASoC: amd: acp: delete unnecessary NULL check
by Dan Carpenter 18 Jul '23

18 Jul '23

The list iterator can't be NULL. Delete the check and pull the code in one tab. Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/amd/acp/acp-rembrandt.c | 22 ++++++++++------------ 1 file changed, 10 insertions(+), 12 deletions(-) diff --git a/sound/soc/amd/acp/acp-rembrandt.c b/sound/soc/amd/acp/acp-rembrandt.c index 21e67ed956d1..cc8284f417c0 100644 --- a/sound/soc/amd/acp/acp-rembrandt.c +++ b/sound/soc/amd/acp/acp-rembrandt.c @@ -266,18 +266,16 @@ static int __maybe_unused rmb_pcm_resume(struct device *dev) acp6x_master_clock_generate(dev); spin_lock(&adata->acp_lock); list_for_each_entry(stream, &adata->stream_list, list) { - if (stream) { - substream = stream->substream; - if (substream && substream->runtime) { - buf_in_frames = (substream->runtime->buffer_size); - buf_size = frames_to_bytes(substream->runtime, buf_in_frames); - config_pte_for_stream(adata, stream); - config_acp_dma(adata, stream, buf_size); - if (stream->dai_id) - restore_acp_i2s_params(substream, adata, stream); - else - restore_acp_pdm_params(substream, adata); - } + substream = stream->substream; + if (substream && substream->runtime) { + buf_in_frames = (substream->runtime->buffer_size); + buf_size = frames_to_bytes(substream->runtime, buf_in_frames); + config_pte_for_stream(adata, stream); + config_acp_dma(adata, stream, buf_size); + if (stream->dai_id) + restore_acp_i2s_params(substream, adata, stream); + else + restore_acp_pdm_params(substream, adata); } } spin_unlock(&adata->acp_lock); -- 2.39.2

2 1

[PATCH 00/11] mfd:: Update Cirrus Logic drivers to use maple tree register caches
by Mark Brown 18 Jul '23

18 Jul '23

With the v6.5 merge window the maple tree register cache has acquired the ability to generate multi-register writes during sync operations, removing the last major limitation it had compared to the rbtree cache. As part of a general modernisation effort update the Cirrus Logic drivers to use the more modern data structure, realistically there is likely to be very little user visible impact either way but generally the implementation choices of the maple tree cache should be a bit better on modern hardware. Signed-off-by: Mark --- Mark Brown (11): mfd: cs47l15: Update to use maple tree register cache mfd: cs47l24: Update to use maple tree register cache mfd: cs47l35: Update to use maple tree register cache mfd: cs47l85: Update to use maple tree register cache mfd: cs47l90: Update to use maple tree register cache mfd: cs47l92: Update to use maple tree register cache mfd: wm5102: Update to use maple tree register cache mfd: wm5110: Update to use maple tree register cache mfd: wm8994: Update to use maple tree register cache mfd: wm8997: Update to use maple tree register cache mfd: wm8998: Update to use maple tree register cache drivers/mfd/cs47l15-tables.c | 8 ++++---- drivers/mfd/cs47l24-tables.c | 2 +- drivers/mfd/cs47l35-tables.c | 8 ++++---- drivers/mfd/cs47l85-tables.c | 8 ++++---- drivers/mfd/cs47l90-tables.c | 8 ++++---- drivers/mfd/cs47l92-tables.c | 8 ++++---- drivers/mfd/wm5102-tables.c | 2 +- drivers/mfd/wm5110-tables.c | 2 +- drivers/mfd/wm8994-regmap.c | 6 +++--- drivers/mfd/wm8997-tables.c | 2 +- drivers/mfd/wm8998-tables.c | 2 +- 11 files changed, 28 insertions(+), 28 deletions(-) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230623-mfd-cirrus-maple-4b3286352baa Best regards, -- Mark Brown <broonie(a)kernel.org>

3 15

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NS70AU
by Werner Sembach 18 Jul '23

18 Jul '23

From: Christoffer Sandberg <cs(a)tuxedo.de> Fixes headset detection on Clevo NS70AU. Co-developed-by: Werner Sembach <wse(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Signed-off-by: Christoffer Sandberg <cs(a)tuxedo.de> Cc: <stable(a)vger.kernel.org> --- 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 e2f8b608de82b..770b2953f0412 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9727,6 +9727,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x5157, "Clevo W517GU1", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x51a1, "Clevo NS50MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x51b1, "Clevo NS50AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x51b3, "Clevo NS70AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.34.1

2 1

X13s volume fixes
by GitHub pull_request - opened 18 Jul '23

18 Jul '23

alsa-project/alsa-ucm-conf pull request #335 was opened from Srinivas-Kandagatla: This patchset fixes some of the Volume issues that are seen on X13s, most of the existing ucm was controlling Digital Gains instead of Analog Gains. With this patchset the volume on both speakers are much better and not saturated. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/335 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/335.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4 00/15] PCI: Define Intel PCI IDs and use them in drivers
by Amadeusz Sławiński 18 Jul '23

18 Jul '23

PCI IDs for Intel HDA are duplicated across quite a few drivers, due to various configurations and historical reasons. Currently almost all uses of HDA PCI IDs have corresponding comment telling which platform it is. Additionally there are some inconsistencies between drivers about which ID corresponds to which device. Simplify things, by adding PCI IDs to global header and make use of them in drivers. This allows for removal of comments by having IDs themselves being self explanatory. Additionally it allows for removal of existing inconsistencies by having one source of truth. Changes from v3: - Fix double space in commit message title (Ilpo) - Comment ACPI/PCI ID handling in Atom SST driver (Andy) - Rebased once again and used --base this time (Andy) Changes from v2: - Change CHV to BSW (Andy) - Fix incorrectly rebased patches (Andy) - Fix commit message and add suggestions from Andy to SST patch (Andy) - Rebased on top of Linus tree commit: 4b810bf037e5 Changes from v1: - Remove BXT-T PCI ID as it is not released (Andy) - Reorder macros to start from pci_match_id() (Andy) - Add comment about BXT->APL name change in commit messages (Andy) - Use SST as part of macro name for DSP only devices (Andy) - Add PCI IDs for all SST combinations (Andy) Changes from RFC: - Sort Intel PCI IDs before adding new ones - Fix ordering of new PCI IDs (Andy) - Define all used Intel IDs (Andy) - Add macros for controller type detection (Andy/Bjorn) - Add set of patches changing to use above macro (Andy/Bjorn) - Use PCI_DEVICE_DATA for Intel IDs in sound/pci/hda/hda_intel.c (Andy) - Commit message wording (Andy) - Remove unnecessary tabs (Andy) Amadeusz Sławiński (15): PCI: Sort Intel PCI IDs by number PCI: Add Intel Audio DSP devices to pci_ids.h ASoC: SOF: Remove unused Broxton PCI ID ALSA: Remove unused Broxton PCI ID ALSA: hda: Add controller matching macros ALSA: hda: Use global PCI match macro ALSA: hda/i915: Use global PCI match macro ASoC: Intel: Skylake: Use global PCI match macro ALSA: intel-dsp-config: Convert to PCI device IDs defines ALSA: hda: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: Skylake: Convert to PCI device IDs defines ASoC: SOF: Intel: Convert to PCI device IDs defines ASoC: Intel: sst: Convert to PCI device IDs defines include/linux/pci_ids.h | 105 +++++-- include/sound/hda_codec.h | 3 - include/sound/hdaudio.h | 26 ++ sound/hda/hdac_i915.c | 7 +- sound/hda/intel-dsp-config.c | 124 ++++---- sound/pci/hda/hda_intel.c | 373 ++++++++++--------------- sound/soc/intel/atom/sst/sst.c | 14 +- sound/soc/intel/atom/sst/sst.h | 7 +- sound/soc/intel/atom/sst/sst_pci.c | 4 +- sound/soc/intel/avs/board_selection.c | 10 +- sound/soc/intel/avs/core.c | 16 +- sound/soc/intel/skylake/skl-messages.c | 16 +- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/intel/skylake/skl.c | 36 +-- sound/soc/sof/intel/pci-apl.c | 8 +- sound/soc/sof/intel/pci-cnl.c | 15 +- sound/soc/sof/intel/pci-icl.c | 12 +- sound/soc/sof/intel/pci-mtl.c | 3 +- sound/soc/sof/intel/pci-skl.c | 6 +- sound/soc/sof/intel/pci-tgl.c | 45 +-- sound/soc/sof/intel/pci-tng.c | 3 +- 21 files changed, 391 insertions(+), 445 deletions(-) base-commit: fdf0eaf11452d72945af31804e2a1048ee1b574c -- 2.34.1

3 20

[PATCH 1/1] ASoC: codec: wm8960: add additional probe check for codec identification
by Frank Li 18 Jul '23

18 Jul '23

The wm8960 codec is not readable, resulting in a NACK for address 0x3d (8-bit). This can partially indicate it. For example: wm8962 codec use the same address but is readable. This additional probe check will help prevent loading the wm8960 module incorrectly on wm8962 hardware. Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- sound/soc/codecs/wm8960.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0d167238a369..783c85912c59 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1415,6 +1415,7 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) struct wm8960_data *pdata = dev_get_platdata(&i2c->dev); struct wm8960_priv *wm8960; int ret; + u8 val; wm8960 = devm_kzalloc(&i2c->dev, sizeof(struct wm8960_priv), GFP_KERNEL); @@ -1436,6 +1437,12 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) else if (i2c->dev.of_node) wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + ret = i2c_master_recv(i2c, &val, sizeof(val)); + if (ret >= 0) { + dev_err(&i2c->dev, "Not wm8960, wm8960 reg can not read by i2c\n"); + return -EINVAL; + } + ret = wm8960_reset(wm8960->regmap); if (ret != 0) { dev_err(&i2c->dev, "Failed to issue reset\n"); -- 2.34.1

3 2

[PATCH 0/3] Add Probe functionality support for amd platforms.
by V sujith kumar Reddy 18 Jul '23

18 Jul '23

The Patch series consists of probe functionality support for amd platforms. V sujith kumar Reddy (3): ASoC: SOF: amd: Add Probe functionality support for amd platforms. ASoC: SOF: Add acp-probe id to sof probe client driver for registration. ASoC: SOF: amd: Add Probe register offset for renoir and rembrandt platform. sound/soc/sof/amd/Kconfig | 8 ++ sound/soc/sof/amd/Makefile | 1 + sound/soc/sof/amd/acp-common.c | 4 + sound/soc/sof/amd/acp-ipc.c | 26 ++++++ sound/soc/sof/amd/acp-probes.c | 147 ++++++++++++++++++++++++++++++ sound/soc/sof/amd/acp.h | 9 ++ sound/soc/sof/amd/pci-rmb.c | 2 + sound/soc/sof/amd/pci-rn.c | 2 + sound/soc/sof/sof-client-probes.c | 1 + 9 files changed, 200 insertions(+) create mode 100644 sound/soc/sof/amd/acp-probes.c -- 2.25.1

2 4

[PATCH] ASoC: fsl_sai: Disable bit clock with transmitter
by Matus Gajdos 18 Jul '23

18 Jul '23

Otherwise bit clock remains running writing invalid data to the DAC. Signed-off-by: Matus Gajdos <matuszpd(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5e09f634c61b..dcc7fbe7acac 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -719,7 +719,7 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir) u32 xcsr, count = 100; regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), - FSL_SAI_CSR_TERE, 0); + FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE, 0); /* TERE will remain set till the end of current frame */ do { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 8254c3547b87..550df87b6a06 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -91,6 +91,7 @@ /* SAI Transmit/Receive Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_SE BIT(30) +#define FSL_SAI_CSR_BCE BIT(28) #define FSL_SAI_CSR_FR BIT(25) #define FSL_SAI_CSR_SR BIT(24) #define FSL_SAI_CSR_xF_SHIFT 16 -- 2.25.1

4 5

[PATCH] ALSA: gus: Fix errors in gus_mem.c
by shijie001＠208suo.com 18 Jul '23

18 Jul '23

The following checkpatch errors are removed: ERROR: trailing whitespace ERROR: "foo * bar" should be "foo *bar" Signed-off-by: Jie Shi <shijie001(a)208suo.com> --- sound/isa/gus/gus_mem.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 3e56c01c4544..00c73c0b068f 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -11,11 +11,11 @@ #include <sound/info.h> #ifdef CONFIG_SND_DEBUG -static void snd_gf1_mem_info_read(struct snd_info_entry *entry, +static void snd_gf1_mem_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer); #endif -void snd_gf1_mem_lock(struct snd_gf1_mem * alloc, int xup) +void snd_gf1_mem_lock(struct snd_gf1_mem *alloc, int xup) { if (!xup) { mutex_lock(&alloc->memory_mutex); @@ -67,7 +67,7 @@ snd_gf1_mem_xalloc(struct snd_gf1_mem *alloc, struct snd_gf1_mem_block *block, return nblock; } -int snd_gf1_mem_xfree(struct snd_gf1_mem * alloc, struct snd_gf1_mem_block * block) +int snd_gf1_mem_xfree(struct snd_gf1_mem *alloc, struct snd_gf1_mem_block *block) { if (block->share) { /* ok.. shared block */ block->share--; @@ -97,7 +97,7 @@ int snd_gf1_mem_xfree(struct snd_gf1_mem * alloc, struct snd_gf1_mem_block * blo return 0; } -static struct snd_gf1_mem_block *snd_gf1_mem_look(struct snd_gf1_mem * alloc, +static struct snd_gf1_mem_block *snd_gf1_mem_look(struct snd_gf1_mem *alloc, unsigned int address) { struct snd_gf1_mem_block *block; @@ -110,7 +110,7 @@ static struct snd_gf1_mem_block *snd_gf1_mem_look(struct snd_gf1_mem * alloc, return NULL; } -static struct snd_gf1_mem_block *snd_gf1_mem_share(struct snd_gf1_mem * alloc, +static struct snd_gf1_mem_block *snd_gf1_mem_share(struct snd_gf1_mem *alloc, unsigned int *share_id) { struct snd_gf1_mem_block *block; @@ -125,8 +125,8 @@ static struct snd_gf1_mem_block *snd_gf1_mem_share(struct snd_gf1_mem * alloc, return NULL; } -static int snd_gf1_mem_find(struct snd_gf1_mem * alloc, - struct snd_gf1_mem_block * block, +static int snd_gf1_mem_find(struct snd_gf1_mem *alloc, + struct snd_gf1_mem_block *block, unsigned int size, int w_16, int align) { struct snd_gf1_bank_info *info = w_16 ? alloc->banks_16 : alloc->banks_8; @@ -177,7 +177,7 @@ static int snd_gf1_mem_find(struct snd_gf1_mem * alloc, return -ENOMEM; } -struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owner, +struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem *alloc, int owner, char *name, int size, int w_16, int align, unsigned int *share_id) { @@ -210,7 +210,7 @@ struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owne return nblock; } -int snd_gf1_mem_free(struct snd_gf1_mem * alloc, unsigned int address) +int snd_gf1_mem_free(struct snd_gf1_mem *alloc, unsigned int address) { int result; struct snd_gf1_mem_block *block; @@ -226,7 +226,7 @@ int snd_gf1_mem_free(struct snd_gf1_mem * alloc, unsigned int address) return -EINVAL; } -int snd_gf1_mem_init(struct snd_gus_card * gus) +int snd_gf1_mem_init(struct snd_gus_card *gus) { struct snd_gf1_mem *alloc; struct snd_gf1_mem_block block; @@ -255,7 +255,7 @@ int snd_gf1_mem_init(struct snd_gus_card * gus) return 0; } -int snd_gf1_mem_done(struct snd_gus_card * gus) +int snd_gf1_mem_done(struct snd_gus_card *gus) { struct snd_gf1_mem *alloc; struct snd_gf1_mem_block *block, *nblock; @@ -271,7 +271,7 @@ int snd_gf1_mem_done(struct snd_gus_card * gus) } #ifdef CONFIG_SND_DEBUG -static void snd_gf1_mem_info_read(struct snd_info_entry *entry, +static void snd_gf1_mem_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_gus_card *gus;

2 1

[PATCH 0/2] Add support for rpmsg sound card on i.MX93 platform
by Chancel Liu 18 Jul '23

18 Jul '23

Support rpmsg sound card on i.MX93 platform. Chancel Liu (2): ASoC: dt-bindings: fsl_rpmsg: Add compatible string for i.MX93 ASoC: fsl_rpmsg: Add support for i.MX93 platform Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml | 1 + sound/soc/fsl/fsl_rpmsg.c | 8 ++++++++ 2 files changed, 9 insertions(+) -- 2.25.1

4 5

[PATCH 0/4] ASoC: qcom: Use the maple tree register cache
by Mark Brown 18 Jul '23

18 Jul '23

The maple tree register cache should now be a good replacement for the rbtree cache in almost all situations, update the Qualcomm CODEC drivers to use the newer cache. --- Mark Brown (4): ASoC: wcd9335: Update to use maple tree register cache ASoC: wcd938x: Update to use maple tree register cache ASoC: wsa881x: Update to use maple tree register cache ASoC: wsa883x: Update to use maple tree register cache sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wcd938x-sdw.c | 2 +- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230629-asoc-qcom-maple-02adff47e26f Best regards, -- Mark Brown <broonie(a)kernel.org>

1 5

[PATCH 1/5] ASoC: bcm: bcm63xx-i2s-whistler: Convert to devm_platform_ioremap_resource()
by Yangtao Li 18 Jul '23

18 Jul '23

Use devm_platform_ioremap_resource() to simplify code. Signed-off-by: Yangtao Li <frank.li(a)vivo.com> --- sound/soc/bcm/bcm63xx-i2s-whistler.c | 16 +--------------- 1 file changed, 1 insertion(+), 15 deletions(-) diff --git a/sound/soc/bcm/bcm63xx-i2s-whistler.c b/sound/soc/bcm/bcm63xx-i2s-whistler.c index 18c51dbbc8dc..c64609718738 100644 --- a/sound/soc/bcm/bcm63xx-i2s-whistler.c +++ b/sound/soc/bcm/bcm63xx-i2s-whistler.c @@ -225,7 +225,6 @@ static int bcm63xx_i2s_dev_probe(struct platform_device *pdev) { int ret = 0; void __iomem *regs; - struct resource *r_mem, *region; struct bcm_i2s_priv *i2s_priv; struct regmap *regmap_i2s; struct clk *i2s_clk; @@ -241,20 +240,7 @@ static int bcm63xx_i2s_dev_probe(struct platform_device *pdev) return PTR_ERR(i2s_clk); } - r_mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r_mem) { - dev_err(&pdev->dev, "Unable to get register resource.\n"); - return -ENODEV; - } - - region = devm_request_mem_region(&pdev->dev, r_mem->start, - resource_size(r_mem), DRV_NAME); - if (!region) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - return -EBUSY; - } - - regs = devm_ioremap_resource(&pdev->dev, r_mem); + regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(regs)) { ret = PTR_ERR(regs); return ret; -- 2.39.0

3 6

[syzbot] [alsa?] memory leak in snd_seq_create_port
by syzbot 18 Jul '23

18 Jul '23

Hello, syzbot found the following issue on: HEAD commit: 3f01e9fed845 Merge tag 'linux-watchdog-6.5-rc2' of git://w.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=14b07344a80000 kernel config: https://syzkaller.appspot.com/x/.config?x=75da4f0a455bdbd3 dashboard link: https://syzkaller.appspot.com/bug?extid=cf8e7fa4eeec59b3d485 compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=15877dc2a80000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=12905004a80000 Downloadable assets: disk image: https://storage.googleapis.com/syzbot-assets/441fb7ea58b8/disk-3f01e9fe.raw… vmlinux: https://storage.googleapis.com/syzbot-assets/8fa7790ba0c3/vmlinux-3f01e9fe.… kernel image: https://storage.googleapis.com/syzbot-assets/5e7a6471dadf/bzImage-3f01e9fe.… IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+cf8e7fa4eeec59b3d485(a)syzkaller.appspotmail.com Warning: Permanently added '10.128.1.1' (ED25519) to the list of known hosts. executing program executing program BUG: memory leak unreferenced object 0xffff888100877000 (size 512): comm "syz-executor257", pid 5012, jiffies 4294941742 (age 12.790s) hex dump (first 32 bytes): 80 ff 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................ 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................ backtrace: [<ffffffff8154bf94>] kmalloc_trace+0x24/0x90 mm/slab_common.c:1076 [<ffffffff83d29e28>] kmalloc include/linux/slab.h:582 [inline] [<ffffffff83d29e28>] kzalloc include/linux/slab.h:703 [inline] [<ffffffff83d29e28>] snd_seq_create_port+0x78/0x300 sound/core/seq/seq_ports.c:135 [<ffffffff83d1f681>] snd_seq_ioctl_create_port+0xe1/0x2a0 sound/core/seq/seq_clientmgr.c:1324 [<ffffffff83d20e5e>] snd_seq_ioctl+0x13e/0x290 sound/core/seq/seq_clientmgr.c:2327 [<ffffffff81685173>] vfs_ioctl fs/ioctl.c:51 [inline] [<ffffffff81685173>] __do_sys_ioctl fs/ioctl.c:870 [inline] [<ffffffff81685173>] __se_sys_ioctl fs/ioctl.c:856 [inline] [<ffffffff81685173>] __x64_sys_ioctl+0x103/0x140 fs/ioctl.c:856 [<ffffffff84a77ff9>] do_syscall_x64 arch/x86/entry/common.c:50 [inline] [<ffffffff84a77ff9>] do_syscall_64+0x39/0xb0 arch/x86/entry/common.c:80 [<ffffffff84c0008b>] entry_SYSCALL_64_after_hwframe+0x63/0xcd BUG: memory leak unreferenced object 0xffff888106742c00 (size 512): comm "syz-executor257", pid 5013, jiffies 4294942276 (age 7.450s) hex dump (first 32 bytes): 80 ff 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................ 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................ backtrace: [<ffffffff8154bf94>] kmalloc_trace+0x24/0x90 mm/slab_common.c:1076 [<ffffffff83d29e28>] kmalloc include/linux/slab.h:582 [inline] [<ffffffff83d29e28>] kzalloc include/linux/slab.h:703 [inline] [<ffffffff83d29e28>] snd_seq_create_port+0x78/0x300 sound/core/seq/seq_ports.c:135 [<ffffffff83d1f681>] snd_seq_ioctl_create_port+0xe1/0x2a0 sound/core/seq/seq_clientmgr.c:1324 [<ffffffff83d20e5e>] snd_seq_ioctl+0x13e/0x290 sound/core/seq/seq_clientmgr.c:2327 [<ffffffff81685173>] vfs_ioctl fs/ioctl.c:51 [inline] [<ffffffff81685173>] __do_sys_ioctl fs/ioctl.c:870 [inline] [<ffffffff81685173>] __se_sys_ioctl fs/ioctl.c:856 [inline] [<ffffffff81685173>] __x64_sys_ioctl+0x103/0x140 fs/ioctl.c:856 [<ffffffff84a77ff9>] do_syscall_x64 arch/x86/entry/common.c:50 [inline] [<ffffffff84a77ff9>] do_syscall_64+0x39/0xb0 arch/x86/entry/common.c:80 [<ffffffff84c0008b>] entry_SYSCALL_64_after_hwframe+0x63/0xcd --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. If the bug is already fixed, let syzbot know by replying with: #syz fix: exact-commit-title If you want syzbot to run the reproducer, reply with: #syz test: git://repo/address.git branch-or-commit-hash If you attach or paste a git patch, syzbot will apply it before testing. If you want to change bug's subsystems, reply with: #syz set subsystems: new-subsystem (See the list of subsystem names on the web dashboard) If the bug is a duplicate of another bug, reply with: #syz dup: exact-subject-of-another-report If you want to undo deduplication, reply with: #syz undup

4 7

[PATCH] ASoC: SOF: ipc3-dtrace: uninitialized data in dfsentry_trace_filter_write()
by Dan Carpenter 17 Jul '23

17 Jul '23

This doesn't check how many bytes the simple_write_to_buffer() writes to the buffer. The only thing that we know is that the first byte is initialized and the last byte of the buffer is set to NUL. However the middle bytes could be uninitialized. There is no need to use simple_write_to_buffer(). This code does not support partial writes but instead passes "pos = 0" as the starting offset regardless of what the user passed as "*ppos". Just use the copy_from_user() function and initialize the whole buffer. Fixes: 671e0b90051e ("ASoC: SOF: Clone the trace code to ipc3-dtrace as fw_tracing implementation") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- The nicer thing to do would have been to use strndup_user() but that comes with a small risk of breaking user space and I can't test this so I didn't do that. sound/soc/sof/ipc3-dtrace.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index 1d3bca2d28dd..35da85a45a9a 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -186,7 +186,6 @@ static ssize_t dfsentry_trace_filter_write(struct file *file, const char __user struct snd_sof_dfsentry *dfse = file->private_data; struct sof_ipc_trace_filter_elem *elems = NULL; struct snd_sof_dev *sdev = dfse->sdev; - loff_t pos = 0; int num_elems; char *string; int ret; @@ -201,11 +200,11 @@ static ssize_t dfsentry_trace_filter_write(struct file *file, const char __user if (!string) return -ENOMEM; - /* assert null termination */ - string[count] = 0; - ret = simple_write_to_buffer(string, count, &pos, from, count); - if (ret < 0) + if (copy_from_user(string, from, count)) { + ret = -EFAULT; goto error; + } + string[count] = '\0'; ret = trace_filter_parse(sdev, string, &num_elems, &elems); if (ret < 0) -- 2.39.2

2 1

[PATCH] ASoC: cs42l51: fix driver to properly autoload with automatic module loading
by Thomas Petazzoni 17 Jul '23

17 Jul '23

In commit 2cb1e0259f50 ("ASoC: cs42l51: re-hook of_match_table pointer"), 9 years ago, some random guy fixed the cs42l51 after it was split into a core part and an I2C part to properly match based on a Device Tree compatible string. However, the fix in this commit is wrong: the MODULE_DEVICE_TABLE(of, ....) is in the core part of the driver, not the I2C part. Therefore, automatic module loading based on module.alias, based on matching with the DT compatible string, loads the core part of the driver, but not the I2C part. And threfore, the i2c_driver is not registered, and the codec is not known to the system, nor matched with a DT node with the corresponding compatible string. In order to fix that, we move the MODULE_DEVICE_TABLE(of, ...) into the I2C part of the driver. The cs42l51_of_match[] array is also moved as well, as it is not possible to have this definition in one file, and the MODULE_DEVICE_TABLE(of, ...) invocation in another file, due to how MODULE_DEVICE_TABLE works. Thanks to this commit, the I2C part of the driver now properly autoloads, and thanks to its dependency on the core part, the core part gets autoloaded as well, resulting in a functional sound card without having to manually load kernel modules. Fixes: 2cb1e0259f50 ("ASoC: cs42l51: re-hook of_match_table pointer") Cc: stable(a)vger.kernel.org Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- sound/soc/codecs/cs42l51-i2c.c | 6 ++++++ sound/soc/codecs/cs42l51.c | 7 ------- sound/soc/codecs/cs42l51.h | 1 - 3 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index b2106ff6a7cb..e7db7bcd0296 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -19,6 +19,12 @@ static struct i2c_device_id cs42l51_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs42l51_i2c_id); +const struct of_device_id cs42l51_of_match[] = { + { .compatible = "cirrus,cs42l51", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs42l51_of_match); + static int cs42l51_i2c_probe(struct i2c_client *i2c) { struct regmap_config config; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index a67cd3ee84e0..a7079ae0ca09 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -823,13 +823,6 @@ int __maybe_unused cs42l51_resume(struct device *dev) } EXPORT_SYMBOL_GPL(cs42l51_resume); -const struct of_device_id cs42l51_of_match[] = { - { .compatible = "cirrus,cs42l51", }, - { } -}; -MODULE_DEVICE_TABLE(of, cs42l51_of_match); -EXPORT_SYMBOL_GPL(cs42l51_of_match); - MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index a79343e8a54e..125703ede113 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -16,7 +16,6 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap); void cs42l51_remove(struct device *dev); int __maybe_unused cs42l51_suspend(struct device *dev); int __maybe_unused cs42l51_resume(struct device *dev); -extern const struct of_device_id cs42l51_of_match[]; #define CS42L51_CHIP_ID 0x1B #define CS42L51_CHIP_REV_A 0x00 -- 2.41.0

2 1

[PATCH v2 0/8] ALSA: emu10k1: add support for high-bitrate modes of E-MU cards
by Oswald Buddenhagen 17 Jul '23

17 Jul '23

This series is what all the work was about: support the "dual-/quad-pumped" modes of the E-MU cards. Oswald Buddenhagen (8): ALSA: emu10k1: introduce alternative E-MU D.A.S. mode ALSA: emu10k1: improve mixer control naming in E-MU D.A.S. mode ALSA: emu10k1: set the "no filtering" bits on PCM voices ALSA: emu10k1: make playback in E-MU D.A.S. mode 32-bit ALSA: add snd_ctl_add_locked() ALSA: emu10k1: add support for 2x/4x word clocks in E-MU D.A.S. mode ALSA: emu10k1: add high-rate capture in E-MU D.A.S. mode ALSA: emu10k1: add high-rate playback in E-MU D.A.S. mode include/sound/control.h | 1 + include/sound/emu10k1.h | 11 + sound/core/control.c | 31 ++ sound/pci/emu10k1/emu10k1.c | 29 +- sound/pci/emu10k1/emu10k1_main.c | 19 +- sound/pci/emu10k1/emufx.c | 109 +++- sound/pci/emu10k1/emumixer.c | 901 +++++++++++++++++++++++++++---- sound/pci/emu10k1/emupcm.c | 422 +++++++++++++-- sound/pci/emu10k1/emuproc.c | 5 + sound/pci/emu10k1/io.c | 30 +- sound/pci/emu10k1/voice.c | 6 + 11 files changed, 1383 insertions(+), 181 deletions(-) Range-diff against v1: 1: c8c1fbba5e52 = 1: 7cb49147164f ALSA: emu10k1: introduce alternative E-MU D.A.S. mode 2: 879066428bee = 2: aa9ead00e045 ALSA: emu10k1: improve mixer control naming in E-MU D.A.S. mode 3: b663229a0f46 = 3: ac830e67322f ALSA: emu10k1: set the "no filtering" bits on PCM voices 4: a9de9a73571e = 4: 26e0b7c02c1d ALSA: emu10k1: make playback in E-MU D.A.S. mode 32-bit 5: 0f1950b03193 ! 5: 94b64e6c123d ALSA: add snd_ctl_add_locked() @@ Commit message This is in fact more symmetrical to snd_ctl_remove() than snd_ctl_add() is - the former could be named snd_ctl_remove_locked() just as well. + One may argue that this is going in the wrong direction, as drivers have + no business managing the lock. This may be true in principle, but in + practice the vast majority of controls is created even before the device + was registered, and therefore before any locking is necessary at all. + That means that an even more radical approach of changing snd_ctl_add() + do do no locking, and converting the call sites that actually need + locking to a new function, would better match reality, and would be + somewhat more efficient at that. However, that seems a bit risky and way + too much work. + This will be used to dynamically change the available controls from another control's put() callback, which is already locked. One might want to add snd_ctl_replace_locked() for completeness, but I have no use for it now. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> + --- + v2: + - extended commit message ## include/sound/control.h ## @@ include/sound/control.h: void snd_ctl_notify_one(struct snd_card * card, unsigned int mask, struct snd_kc 6: 511efe4ac7ad ! 6: 7bc314bae7f2 ALSA: emu10k1: add support for 2x/4x word clocks in E-MU D.A.S. mode @@ Commit message on playback, and throw away the excess ones on capture. Input-to-output monitoring does actually use the full sample rate, though. - Notably, add_ctls() now uses snd_ctl_add_locked(), so it doesn't - deadlock when called from snd_emu1010_clock_shift_put(). This also - affects the initial creation of the controls, which is OK, as that is - done before the card is registered, so no concurrent access can occur. + Due to hardware constraints, changing the clock multiplier (CM) changes + the available audio ports and the number of available channels. This has + an impact on the channel routing mixer controls. One way to deal with + this would be presenting a union of all possibilities, and simply + ignoring currently inapplicable settings. However, this would be a + terrible user experience, and go against the spirit of prior patches + aimed at decluttering the mixer. Therefore, we do dynamic + reconfiguration (DR) of the mixer in response to changing the CM. + + DR is somewhat controversial, as it has the potential to crash poorly + programmed applications. But that in itself isn't a very convincing + argument against it, as by that logic we'd have to ban all hot-plugging. + Such crashes would also not really qualify as regressions, as the D.A.S. + mode is a new opt-in feature, and therefore no previously stable setups + would be impacted. Also, pendantically, the driver already had DR via + SNDRV_EMU10K1_IOCTL_CODE_POKE. A somewhat valid concern is that changing + mixer settings is a non-privileged operation and therefore potential + crashes could be exploited for a somewhat more impactful nuisance attack + on another user than messing with the mixer per se. However, systemd & + co. limit device access to the user currently logged in on the seat + owning the device. + + There is a specific concern about doing DR in response to changing a + mixer control's value, as an application may legitimately react to DR by + updating all mixer settings in turn. However, that update should write + the same value to the clock multiplier, thus terminating the recursion. + + One may limit DR to merely disabling inapplicable controls, in the hope + that this would be better handled than completely tearing down and + re-creating controls as we do. However, there is no guarantee for that. + And because it is impossible to disable particular enum values within a + control, it would be necessary to have three complete sets of + per-channel controls. This would yield an extremely cluttered and + confusing UI if the application (reasonably) chose to merely visually + disable inactive controls rather than hiding them. + + We do the DR synchronously from within snd_emu1010_clock_shift_put(). + This was enabled by commit 5bbb1ab5bd ("control: use counting semaphore + as write lock for ELEM_WRITE operation"); we merely need to make + add_ctls() use snd_ctl_add_locked() instead of snd_ctl_add(), so it + doesn't deadlock. That also affects the initial creation of the + controls, which is OK, as that is done before the card is registered, so + no concurrent access can occur. + + It would be possible to do the DR in a tasklet after the ioctl finishes. + However, it is not obvious what actual problem that would solve, and the + added asynchronicity would significantly complicate matters, esp. wrt. + the batch updates expected during mixer state restoration. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> + --- + v2: + - expanded commit message ## include/sound/emu10k1.h ## @@ include/sound/emu10k1.h: struct snd_emu1010 { 7: d5cb50ca707f = 7: 72a156fb32cd ALSA: emu10k1: add high-rate capture in E-MU D.A.S. mode 8: 319425a4ccb6 ! 8: 6d35891832b3 ALSA: emu10k1: add high-rate playback in E-MU D.A.S. mode @@ Commit message d948035a92 ("Remove PCM xfer_align sw params"). Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> + --- + v2: + - fixed `sparse` warning re missing __user annotation ## sound/pci/emu10k1/emufx.c ## @@ sound/pci/emu10k1/emufx.c: static int _snd_emu10k1_das_init_efx(struct snd_emu10k1 *emu) @@ sound/pci/emu10k1/emupcm.c: static int snd_emu10k1_efx_playback_trigger(struct s + for (i = 0; i < channels; i++) { + for (j = 0; j < subchans; j++) { + u32 *dst = get_dma_ptr_x(runtime, shift, i, j, hwoff); -+ u32 *src = (u32 *)buf + j * channels + i; ++ u32 __user *src = (u32 __user *)buf + j * channels + i; + for (k = 0; k < frames; k++, dst++, src += voices) + unsafe_get_user(*dst, src, faulted); + } -- 2.40.0.152.g15d061e6df

2 14

[PATCH 0/2] Fix CSC3551 speaker sound problem for machines without a valid ACPI _DSD
by David Xu 17 Jul '23

17 Jul '23

As the comments added in commit 4d4c4bff4f8ed79d95e05 ("ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties"), CSC3551 requires a valid _DSD to work and the current implementation just fails when no _DSD can be found for CSC3551. However it is a fact that many OEMs hardcoded the configurations needed by CSC3551 into their proprietary software for various 2022 and later laptop models, and this makes the Linux installations on these models cannot make any speaker sound. Meanwhile, at this point of time, we see no hope that these OEMs would ever fix this problem via a BIOS update. To address the problem, this patch series contains two patches: Patch 1 for cs35l41 hda driver: a fixup mechanism is introduced that when the driver found there is no valid _DSD that contains the configurations, a fixup function would try to find a fixup entry that contains a proper cs35l41 configuration from a pre-defined fixup table by matching the CSC3551 ACPI _SUB id. If found, the fixup function would apply the cs35l41 configurations retrived from the entry. In this patch the fixup table only contains some entries for three Lenovo laptop models: namely 16IAH7, 16IAX7 and 16ARHA7. However as is known, several other laptop models from ASUS and HP also suffer from this no valid _DSD problem and could have it addressed with this fixup mechanism when proper fixup entries are inserted. Patch 2 for realtek hda driver: add quirks for Lenovo 16IAH7, 16IAX7 and 16ARHA7 so that cs35l41 playback hook could be registered. Please note that for these quirks to work patch 1 has to be applied. David Xu (2): ALSA: hda: cs35l41: Add fixups for machines without a valid ACPI _DSD ALSA: hda/realtek: Add quirks for Lenovo 16IAH7, 16IAX7 and 16ARHA7 sound/pci/hda/cs35l41_hda.c | 160 ++++++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 5 ++ 2 files changed, 165 insertions(+) -- 2.41.0

3 4

[PATCH] ALSA: emu10k1: do away with extra voices in E-MU D.A.S. mode
by Oswald Buddenhagen 17 Jul '23

17 Jul '23

Maybe somewhat unsurprisingly, Creative Inc. learned from the period interrupt disaster of the EMU10K1, and equipped Audigy with a way to delay the interrupts by a configurable amount. So make use of that. This allows us to do away with the extra voices for driving interrupts. The most salient advantage of that is that we can maximize the number of playback channels in 24-bit 192 kHz mode, bringing it up to 8. A minor point is that no bandwidth is wasted on fetching data that is discarded anyway. The removal of the extra voices has one disadvantage: the hardware is limited to two periods, thus increasing the likelihood of underruns. Therefore, we do this only in D.A.S. mode, which aims at high-quality low-latency multi-channel recording, and therefore can be expected to be operated in an optimal environment. Also, prior to a few commits back, the multi-channel playback was actually limited to two periods anyway (for no good reason). Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- this is obviously meaningless without the d.a.s. series. i'm posting it only so it's on record. --- This patch needs to be applied on top of the patch titled "ALSA: emu10k1: add high-rate playback in E-MU D.A.S. mode". --- include/sound/emu10k1.h | 11 +++++++ sound/pci/emu10k1/emu10k1_main.c | 16 ++++++++++ sound/pci/emu10k1/emupcm.c | 52 +++++++++++++++++++++++--------- sound/pci/emu10k1/irq.c | 8 +++++ 4 files changed, 73 insertions(+), 14 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index fd4cf7d6eb3f..0931b1e65e56 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -838,7 +838,18 @@ SUB_REG(FXIDX, IDX, 0x0000ffff) #define A_SPRA 0x6b /* S/PDIF Host Record Address */ #define A_SPRC 0x6c /* S/PDIF Host Record Control */ +// NOTE: 64-bit +// Voice 0 holds the counter, while voice 1 holds the enable bits. +// Bits in IPR are set according to incoming IRQs as usual. +// When a corresponding bit is set in the delay enable register, +// the counter starts, and the IRQ isn't delivered until timeout. +// Additional delayed IRQs arriving during the countdown do not +// retrigger it, so they will be delayed less or not at all. +// Non-delayed IRQs arriving during the countdown are delivered +// immediately, while the countdown continues. #define A_DICE 0x6d /* Delayed Interrupt Counter & Enable */ +#define A_DICE_DELAY_MASK 0x000003ff /* Interrupt delay minus 1 */ +#define A_DICE_COUNTER_MASK 0x03ff0000 /* Counter's current value */ #define A_TTB 0x6e /* Tank Table Base */ #define A_TDOF 0x6f /* Tank Delay Offset */ diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 2125925c2d5e..97a4be6eaf7a 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -159,6 +159,8 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir) /* disable channel interrupt */ CLIEL, 0, CLIEH, 0, + HLIEL, 0, + HLIEH, 0, /* disable stop on loop end */ SOLEL, 0, @@ -174,6 +176,12 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir) AC97SLOT_REAR_LEFT); } + if (emu->das_mode) { + // Enable delayed period interrupts + snd_emu10k1_ptr_write(emu, A_DICE, 0, 64 - 3 + 1); + snd_emu10k1_ptr_write(emu, A_DICE, 1, IPR_CHANNELLOOP); + } + /* init envelope engine */ for (ch = 0; ch < NUM_G; ch++) snd_emu10k1_voice_init(emu, ch); @@ -414,6 +422,12 @@ int snd_emu10k1_done(struct snd_emu10k1 *emu) else snd_emu10k1_ptr_write(emu, DBG, 0, EMU10K1_DBG_SINGLE_STEP); + if (emu->das_mode) { + // Disable delayed period interrupts + snd_emu10k1_ptr_write(emu, A_DICE, 0, 0); + snd_emu10k1_ptr_write(emu, A_DICE, 1, 0); + } + snd_emu10k1_ptr_write_multiple(emu, 0, /* reset recording buffers */ MICBS, 0, @@ -429,6 +443,8 @@ int snd_emu10k1_done(struct snd_emu10k1 *emu) /* disable channel interrupt */ CLIEL, 0, CLIEH, 0, + HLIEL, 0, + HLIEH, 0, SOLEL, 0, SOLEH, 0, diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index ccb0e13da3dd..b7841076f019 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -96,7 +96,9 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm *epcm, if (err < 0) return err; - if (epcm->extra == NULL) { + if (epcm->emu->das_mode) { + epcm->voices[0]->interrupt = snd_emu10k1_pcm_interrupt; + } else if (epcm->extra == NULL) { // The hardware supports only (half-)loop interrupts, so to support an // arbitrary number of periods per buffer, we use an extra voice with a // period-sized loop as the interrupt source. Additionally, the interrupt @@ -500,13 +502,8 @@ static int snd_emu10k1_efx_playback_prepare(struct snd_pcm_substream *substream) epcm->pitch_target = PITCH_48000; start_addr = epcm->start_addr >> 1; // 16-bit voices - - extra_size = runtime->period_size >> shift; channel_size = runtime->buffer_size >> shift; - snd_emu10k1_pcm_init_extra_voice(emu, epcm->extra, true, - start_addr, start_addr + extra_size); - if (das_mode) { unsigned count = 1 << shift; start_addr >>= 1; @@ -520,6 +517,10 @@ static int snd_emu10k1_efx_playback_prepare(struct snd_pcm_substream *substream) } } } else { + extra_size = runtime->period_size >> shift; + snd_emu10k1_pcm_init_extra_voice(emu, epcm->extra, true, + start_addr, start_addr + extra_size); + epcm->ccca_start_addr = start_addr; for (i = 0; i < runtime->channels; i++) { snd_emu10k1_pcm_init_voices(emu, epcm->voices[i], true, false, @@ -547,7 +548,7 @@ static const struct snd_pcm_hardware snd_emu10k1_efx_playback = .buffer_bytes_max = (128*1024), .period_bytes_max = (128*1024), .periods_min = 2, - .periods_max = 1024, + .periods_max = 2, .fifo_size = 0, }; @@ -698,8 +699,16 @@ static void snd_emu10k1_playback_prepare_voices(struct snd_emu10k1 *emu, // This is why all other (open) drivers for these chips use timer-based // interrupts. // - eloop_start += (epcm->resume_pos + eloop_size - 3) % eloop_size; - snd_emu10k1_ptr_write(emu, CCCA_CURRADDR, epcm->extra->number, eloop_start); + // But then, Audigy introduced delayed interrupt functionality, so we + // use it in D.A.S. mode, where we may need all voices. We can't use + // it in regular mode due to supporting multiple substreams, which the + // delayed IRQs cannot really handle; also, we prefer supporting an + // arbitrary number of periods there, for which we need the extra voice. + // + if (!emu->das_mode) { + eloop_start += (epcm->resume_pos + eloop_size - 3) % eloop_size; + snd_emu10k1_ptr_write(emu, CCCA_CURRADDR, epcm->extra->number, eloop_start); + } // It takes a moment until the cache fills complete, // but the unmuting takes long enough for that. @@ -793,13 +802,25 @@ static void snd_emu10k1_playback_set_running(struct snd_emu10k1 *emu, struct snd_emu10k1_pcm *epcm) { epcm->running = 1; - snd_emu10k1_voice_intr_enable(emu, epcm->extra->number); + if (emu->das_mode) { + unsigned int voice = epcm->voices[0]->number; + snd_emu10k1_voice_half_loop_intr_enable(emu, voice); + snd_emu10k1_voice_intr_enable(emu, voice); + } else { + snd_emu10k1_voice_intr_enable(emu, epcm->extra->number); + } } static void snd_emu10k1_playback_set_stopped(struct snd_emu10k1 *emu, struct snd_emu10k1_pcm *epcm) { - snd_emu10k1_voice_intr_disable(emu, epcm->extra->number); + if (emu->das_mode) { + unsigned int voice = epcm->voices[0]->number; + snd_emu10k1_voice_half_loop_intr_disable(emu, voice); + snd_emu10k1_voice_intr_disable(emu, voice); + } else { + snd_emu10k1_voice_intr_disable(emu, epcm->extra->number); + } epcm->running = 0; } @@ -1064,7 +1085,8 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, result = snd_emu10k1_voice_clear_loop_stop_multiple_atomic(emu, mask); if (result == 0) { // The extra voice is allowed to lag a bit - snd_emu10k1_playback_trigger_voice(emu, epcm->extra); + if (!emu->das_mode) + snd_emu10k1_playback_trigger_voice(emu, epcm->extra); goto leave; } @@ -1080,7 +1102,8 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - snd_emu10k1_playback_stop_voice(emu, epcm->extra); + if (!emu->das_mode) + snd_emu10k1_playback_stop_voice(emu, epcm->extra); snd_emu10k1_efx_playback_stop_voices( emu, epcm, das_mode, count, runtime->channels); @@ -1428,15 +1451,16 @@ static int snd_emu10k1_efx_playback_open(struct snd_pcm_substream *substream) SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_PAUSE; if (shift == 2) - runtime->hw.channels_max = 7; // FIXME: should be 8, but extra voice ... + runtime->hw.channels_max = 8; err = snd_pcm_hw_constraint_step( runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 1 << shift); if (err < 0) { kfree(epcm); return err; } } runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE; + runtime->hw.periods_max = 2; // Not using an extra voice } } err = snd_emu10k1_playback_set_constraints(emu, runtime); diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c index 71aa90b9cc88..be2dbed12e74 100644 --- a/sound/pci/emu10k1/irq.c +++ b/sound/pci/emu10k1/irq.c @@ -23,6 +23,14 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id) "Suspected sound card removal\n"); break; } + if ((status & IPR_CHANNELLOOP) && emu->das_mode && + (snd_emu10k1_ptr_read(emu, A_DICE, 0) & A_DICE_COUNTER_MASK)) { + // If another IRQ "punched through" the delay, + // we mask away the delayed IRQ. + status &= ~(IPR_CHANNELLOOP | IPR_CHANNELNUMBERMASK); + if (!status) // Will usually happen in 2nd iteration + break; + } if (++timeout == 1000) { dev_info(emu->card->dev, "emu10k1 irq routine failure\n"); break; base-commit: 3c04ccbdf86e7e93e06e68f1915e39eb55afb0d0 prerequisite-patch-id: 45677e4f73ea654bda3258c1143c8027173e4b5c prerequisite-patch-id: 358302d653684999fa1d7b7c0b9c643d6daf768d prerequisite-patch-id: e5ad839d4411bacb9ed5d639a0bf151a3b9dfeb1 prerequisite-patch-id: aa2b589f883089a61a979b76897a60865ab2fd8a prerequisite-patch-id: 86710b3c986270dcdbabbbbad9f0d86ed634a3ad prerequisite-patch-id: 7d3ce3deda4c9a317868cc13150b3ece13f0096e prerequisite-patch-id: 3bd887e1980015ed352cc20eea63b69f3cf8a9c2 -- 2.40.0.152.g15d061e6df

1 0

[PATCH 1/2] ALSA: emu10k1: clean up driver status comments
by Oswald Buddenhagen 17 Jul '23

17 Jul '23

Empty BUGS and TODO sections don't really help anyone, so remove them. Version information is chronically outdated, and not really useful in a git world anyway, so remove it as well. Also remove duplicated (and outdated, of course) status section from p16v.h (the one in p16v.c is in better shape). Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emu10k1_main.c | 6 ---- sound/pci/emu10k1/emufx.c | 6 ---- sound/pci/emu10k1/emumixer.c | 6 ---- sound/pci/emu10k1/emupcm.c | 6 ---- sound/pci/emu10k1/emuproc.c | 6 ---- sound/pci/emu10k1/io.c | 6 ---- sound/pci/emu10k1/irq.c | 6 ---- sound/pci/emu10k1/p16v.h | 56 -------------------------------- sound/pci/emu10k1/p17v.h | 1 - sound/pci/emu10k1/timer.c | 6 ---- sound/pci/emu10k1/tina2.h | 1 - sound/pci/emu10k1/voice.c | 6 ---- 12 files changed, 112 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index a11fcba4b9af..529d0bba87da 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -8,12 +8,6 @@ * Added support for Audigy 2 Value. * Added EMU 1010 support. * General bug fixes and enhancements. - * - * BUGS: - * -- - * - * TODO: - * -- */ #include <linux/sched.h> diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 9904bcfee106..e0ad339c5bbf 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -6,12 +6,6 @@ * * Copyright (c) by James Courtier-Dutton <James(a)superbug.co.uk> * Added EMU 1010 support. - * - * BUGS: - * -- - * - * TODO: - * -- */ #include <linux/pci.h> diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 9a94f08f2463..6ff59c44da36 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -8,12 +8,6 @@ * * Copyright (c) by James Courtier-Dutton <James(a)superbug.co.uk> * Added EMU 1010 support. - * - * BUGS: - * -- - * - * TODO: - * -- */ #include <linux/time.h> diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 8b3d1b35d6e7..7a1d448f0656 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -4,12 +4,6 @@ * Creative Labs, Inc. * Routines for control of EMU10K1 chips / PCM routines * Multichannel PCM support Copyright (c) Lee Revell <rlrevell(a)joe-job.com> - * - * BUGS: - * -- - * - * TODO: - * -- */ #include <linux/pci.h> diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 5533277e4d47..39b422297c70 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -6,12 +6,6 @@ * * Copyright (c) by James Courtier-Dutton <James(a)superbug.co.uk> * Added EMU 1010 support. - * - * BUGS: - * -- - * - * TODO: - * -- */ #include <linux/slab.h> diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index c695cb863e5e..7e4483c5bd2a 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -3,12 +3,6 @@ * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> * Creative Labs, Inc. * Routines for control of EMU10K1 chips - * - * BUGS: - * -- - * - * TODO: - * -- */ #include <linux/time.h> diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c index 8573248dd799..71aa90b9cc88 100644 --- a/sound/pci/emu10k1/irq.c +++ b/sound/pci/emu10k1/irq.c @@ -3,12 +3,6 @@ * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> * Creative Labs, Inc. * Routines for IRQ control of EMU10K1 chips - * - * BUGS: - * -- - * - * TODO: - * -- */ #include <linux/time.h> diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h index 9d429ad1feff..95ab8071751b 100644 --- a/sound/pci/emu10k1/p16v.h +++ b/sound/pci/emu10k1/p16v.h @@ -2,62 +2,6 @@ /* * Copyright (c) by James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver p16v chips - * Version: 0.21 - * - * FEATURES currently supported: - * Output fixed at S32_LE, 2 channel to hw:0,0 - * Rates: 44.1, 48, 96, 192. - * - * Changelog: - * 0.8 - * Use separate card based buffer for periods table. - * 0.9 - * Use 2 channel output streams instead of 8 channel. - * (8 channel output streams might be good for ASIO type output) - * Corrected speaker output, so Front -> Front etc. - * 0.10 - * Fixed missed interrupts. - * 0.11 - * Add Sound card model number and names. - * Add Analog volume controls. - * 0.12 - * Corrected playback interrupts. Now interrupt per period, instead of half period. - * 0.13 - * Use single trigger for multichannel. - * 0.14 - * Mic capture now works at fixed: S32_LE, 96000Hz, Stereo. - * 0.15 - * Force buffer_size / period_size == INTEGER. - * 0.16 - * Update p16v.c to work with changed alsa api. - * 0.17 - * Update p16v.c to work with changed alsa api. Removed boot_devs. - * 0.18 - * Merging with snd-emu10k1 driver. - * 0.19 - * One stereo channel at 24bit now works. - * 0.20 - * Added better register defines. - * 0.21 - * Split from p16v.c - * - * BUGS: - * Some stability problems when unloading the snd-p16v kernel module. - * -- - * - * TODO: - * SPDIF out. - * Find out how to change capture sample rates. E.g. To record SPDIF at 48000Hz. - * Currently capture fixed at 48000Hz. - * - * -- - * GENERAL INFO: - * Model: SB0240 - * P16V Chip: CA0151-DBS - * Audigy 2 Chip: CA0102-IAT - * AC97 Codec: STAC 9721 - * ADC: Philips 1361T (Stereo 24bit) - * DAC: CS4382-K (8-channel, 24bit, 192Khz) * * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes(a)nc.rr.com> diff --git a/sound/pci/emu10k1/p17v.h b/sound/pci/emu10k1/p17v.h index d4ada1c430c8..ee4f4ab4b79c 100644 --- a/sound/pci/emu10k1/p17v.h +++ b/sound/pci/emu10k1/p17v.h @@ -2,7 +2,6 @@ /* * Copyright (c) by James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver p17v chips - * Version: 0.01 */ /******************************************************************************/ diff --git a/sound/pci/emu10k1/timer.c b/sound/pci/emu10k1/timer.c index f3c78adf3248..8798604e7f98 100644 --- a/sound/pci/emu10k1/timer.c +++ b/sound/pci/emu10k1/timer.c @@ -3,12 +3,6 @@ * Copyright (c) by Lee Revell <rlrevell(a)joe-job.com> * Clemens Ladisch <clemens(a)ladisch.de> * Routines for control of EMU10K1 chips - * - * BUGS: - * -- - * - * TODO: - * -- */ #include <linux/time.h> diff --git a/sound/pci/emu10k1/tina2.h b/sound/pci/emu10k1/tina2.h index 7fd235345292..e3fcb290271c 100644 --- a/sound/pci/emu10k1/tina2.h +++ b/sound/pci/emu10k1/tina2.h @@ -2,7 +2,6 @@ /* * Copyright (c) by James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver tina2 chips - * Version: 0.1 */ /********************************************************************************************************/ diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c index 6939498e26f0..ffe87f359a0e 100644 --- a/sound/pci/emu10k1/voice.c +++ b/sound/pci/emu10k1/voice.c @@ -6,12 +6,6 @@ * Routines for control of EMU10K1 chips - voice manager * * Rewrote voice allocator for multichannel support - rlrevell 12/2004 - * - * BUGS: - * -- - * - * TODO: - * -- */ #include <linux/time.h> -- 2.40.0.152.g15d061e6df

2 2

[PATCH v3] ALSA: emu10k1: set the "no filtering" bits on PCM voices on Audigy
by Oswald Buddenhagen 17 Jul '23

17 Jul '23

Given that the filter is already set to neutral for PCM voices, the only observable effect is that the Z1/Z2/FXBUS registers don't have a stray bit set for negative numbers anymore. The bit is below the ones significant for output, but it would mess with 32-bit sample recombination, which we intend to add. kX-project does that, but I had to figure out myself why. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- v3: - simplify, as this is actually unrelated to the interpolator --- include/sound/emu10k1.h | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 43c097952c3c..ef948e45b4c6 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -902,6 +902,11 @@ SUB_REG_NC(A_EHC, A_I2S_CAPTURE_RATE, 0x00000e00) /* This sets the capture PCM #define A_TTDA 0x7a /* Tank Table DMA Address */ #define A_TTDD 0x7b /* Tank Table DMA Data */ +// In A_FXRT1 & A_FXRT2, the 0x80 bit of each byte completely disables the +// filter (CVCF_CURRENTFILTER) for the corresponding channel. There is no +// effect on the volume (CVCF_CURRENTVOLUME) or the interpolator's filter +// (CCCA_INTERPROM_MASK). + #define A_FXRT2 0x7c #define A_FXRT_CHANNELE 0x0000003f /* Effects send bus number for channel's effects send E */ #define A_FXRT_CHANNELF 0x00003f00 /* Effects send bus number for channel's effects send F */ @@ -914,8 +919,6 @@ SUB_REG_NC(A_EHC, A_I2S_CAPTURE_RATE, 0x00000e00) /* This sets the capture PCM #define A_FXSENDAMOUNT_G_MASK 0x0000FF00 #define A_FXSENDAMOUNT_H_MASK 0x000000FF -/* 0x7c, 0x7e "high bit is used for filtering" */ - /* The send amounts for this one are the same as used with the emu10k1 */ #define A_FXRT1 0x7e #define A_FXRT_CHANNELA 0x0000003f @@ -1523,10 +1526,10 @@ struct snd_emu10k1_pcm_mixer { ((route[0] | (route[1] << 4) | (route[2] << 8) | (route[3] << 12)) << 16) #define snd_emu10k1_compose_audigy_fxrt1(route) \ -((unsigned int)route[0] | ((unsigned int)route[1] << 8) | ((unsigned int)route[2] << 16) | ((unsigned int)route[3] << 24)) +((unsigned int)route[0] | ((unsigned int)route[1] << 8) | ((unsigned int)route[2] << 16) | ((unsigned int)route[3] << 24) | 0x80808080) #define snd_emu10k1_compose_audigy_fxrt2(route) \ -((unsigned int)route[4] | ((unsigned int)route[5] << 8) | ((unsigned int)route[6] << 16) | ((unsigned int)route[7] << 24)) +((unsigned int)route[4] | ((unsigned int)route[5] << 8) | ((unsigned int)route[6] << 16) | ((unsigned int)route[7] << 24) | 0x80808080) #define snd_emu10k1_compose_audigy_sendamounts(vol) \ (((unsigned int)vol[4] << 24) | ((unsigned int)vol[5] << 16) | ((unsigned int)vol[6] << 8) | (unsigned int)vol[7]) -- 2.40.0.152.g15d061e6df

2 1

[PATCH v2] ALSA: emu10k1: track loss of external clock on E-MU cards
by Oswald Buddenhagen 17 Jul '23

17 Jul '23

This uses IRQs to track spontaneous changes to the word clock source register. FWIW, that this can happen in the first place is the reason why it is futile to lock the clock source mixer setting while the device is open - we can't consistently control the rate anyway. Though arguably, we should reset any open streams when that happens, as they become corrupted anyway. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- v2: - use driver-local lock instead of core's control_rwsem --- FIXME? while i'm not sure, i think this won't notice seamless switches between 44.1 and 48 kHz. assuming that's the case, this seems like a minor issue: firstly, just about nothing actually produces such a seamless switch - my only device that can even do that is the e-mu card itself, but the driver disrupts that by temporarily muting the output. secondly, the user is unlikely to select an external source before setting it up properly. and the easy workaround is actually never doing that. regardless, to actually test that i'd need a second e-mu card. --- include/sound/emu10k1.h | 5 +++++ sound/pci/emu10k1/emu10k1.c | 1 + sound/pci/emu10k1/emu10k1_main.c | 30 +++++++++++++++++++++++++++++- sound/pci/emu10k1/emumixer.c | 12 ++++++++---- 4 files changed, 43 insertions(+), 5 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 43c097952c3c..7c55a8244747 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -992,6 +992,9 @@ SUB_REG_NC(A_EHC, A_I2S_CAPTURE_RATE, 0x00000e00) /* This sets the capture PCM #define EMU_HANA_WCLOCK_4X 0x10 #define EMU_HANA_WCLOCK_MULT_RESERVED 0x18 +// If the selected external clock source is/becomes invalid or incompatible +// with the clock multiplier, the clock source is reset to this value, and +// a WCLK_CHANGED interrupt is raised. #define EMU_HANA_DEFCLOCK 0x06 /* 000000x 1 bits Default Word Clock */ #define EMU_HANA_DEFCLOCK_48K 0x00 #define EMU_HANA_DEFCLOCK_44_1K 0x01 @@ -1679,6 +1682,7 @@ struct snd_emu1010 { unsigned int optical_in; /* 0:SPDIF, 1:ADAT */ unsigned int optical_out; /* 0:SPDIF, 1:ADAT */ struct work_struct firmware_work; + struct work_struct clock_work; }; struct snd_emu10k1 { @@ -1753,6 +1757,7 @@ struct snd_emu10k1 { struct snd_kcontrol *ctl_efx_send_routing; struct snd_kcontrol *ctl_efx_send_volume; struct snd_kcontrol *ctl_efx_attn; + struct snd_kcontrol *ctl_clock_source; void (*hwvol_interrupt)(struct snd_emu10k1 *emu, unsigned int status); void (*capture_interrupt)(struct snd_emu10k1 *emu, unsigned int status); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 1a13c086e86c..421053569aa0 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -192,6 +192,7 @@ static int snd_emu10k1_suspend(struct device *dev) emu->suspend = 1; cancel_work_sync(&emu->emu1010.firmware_work); + cancel_work_sync(&emu->emu1010.clock_work); snd_ac97_suspend(emu->ac97); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index a11fcba4b9af..604645bfc908 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -789,6 +789,30 @@ static void emu1010_firmware_work(struct work_struct *work) } } +static void emu1010_clock_work(struct work_struct *work) +{ + struct snd_emu10k1 *emu; + struct snd_ctl_elem_id id; + + emu = container_of(work, struct snd_emu10k1, + emu1010.clock_work); + if (emu->card->shutdown) + return; +#ifdef CONFIG_PM_SLEEP + if (emu->suspend) + return; +#endif + + spin_lock_irq(&emu->reg_lock); + // This is the only thing that can actually happen. + emu->emu1010.clock_source = emu->emu1010.clock_fallback; + emu->emu1010.wclock = 1 - emu->emu1010.clock_source; + snd_emu1010_update_clock(emu); + spin_unlock_irq(&emu->reg_lock); + snd_ctl_build_ioff(&id, emu->ctl_clock_source, 0); + snd_ctl_notify(emu->card, SNDRV_CTL_EVENT_MASK_VALUE, &id); +} + static void emu1010_interrupt(struct snd_emu10k1 *emu) { u32 sts; @@ -802,6 +826,8 @@ static void emu1010_interrupt(struct snd_emu10k1 *emu) } else if (sts & EMU_HANA_IRQ_DOCK) { schedule_work(&emu->emu1010.firmware_work); } + if (sts & EMU_HANA_IRQ_WCLK_CHANGED) + schedule_work(&emu->emu1010.clock_work); } /* @@ -901,7 +927,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) emu->gpio_interrupt = emu1010_interrupt; // Note: The Audigy INTE is set later snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, - EMU_HANA_IRQ_DOCK | EMU_HANA_IRQ_DOCK_LOST); + EMU_HANA_IRQ_DOCK | EMU_HANA_IRQ_DOCK_LOST | EMU_HANA_IRQ_WCLK_CHANGED); snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg); // Clear pending IRQs emu->emu1010.clock_source = 1; /* 48000 */ @@ -943,6 +969,7 @@ static void snd_emu10k1_free(struct snd_card *card) snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); } cancel_work_sync(&emu->emu1010.firmware_work); + cancel_work_sync(&emu->emu1010.clock_work); release_firmware(emu->firmware); release_firmware(emu->dock_fw); snd_util_memhdr_free(emu->memhdr); @@ -1522,6 +1549,7 @@ int snd_emu10k1_create(struct snd_card *card, emu->synth = NULL; emu->get_synth_voice = NULL; INIT_WORK(&emu->emu1010.firmware_work, emu1010_firmware_work); + INIT_WORK(&emu->emu1010.clock_work, emu1010_clock_work); /* read revision & serial */ emu->revision = pci->revision; pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial); diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 9a94f08f2463..f39025748072 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -986,17 +986,21 @@ static int snd_emu1010_clock_source_put(struct snd_kcontrol *kcontrol, val = ucontrol->value.enumerated.item[0] ; if (val >= emu_ci->num) return -EINVAL; + spin_lock_irq(&emu->reg_lock); change = (emu->emu1010.clock_source != val); if (change) { emu->emu1010.clock_source = val; emu->emu1010.wclock = emu_ci->vals[val]; + snd_emu1010_update_clock(emu); snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE); snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, emu->emu1010.wclock); + spin_unlock_irq(&emu->reg_lock); + msleep(10); // Allow DLL to settle snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); - - snd_emu1010_update_clock(emu); + } else { + spin_unlock_irq(&emu->reg_lock); } return change; } @@ -2336,8 +2340,8 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, emu1010_map_source(emu_ri, emu_ri->out_dflts[i]); snd_emu1010_apply_sources(emu); - err = snd_ctl_add(card, - snd_ctl_new1(&snd_emu1010_clock_source, emu)); + kctl = emu->ctl_clock_source = snd_ctl_new1(&snd_emu1010_clock_source, emu); + err = snd_ctl_add(card, kctl); if (err < 0) return err; err = snd_ctl_add(card, -- 2.40.0.152.g15d061e6df

2 1

[PATCH] soundwire: Explicitly include correct DT includes
by Rob Herring 17 Jul '23

17 Jul '23

The DT of_device.h and of_platform.h date back to the separate of_platform_bus_type before it as merged into the regular platform bus. As part of that merge prepping Arm DT support 13 years ago, they "temporarily" include each other. They also include platform_device.h and of.h. As a result, there's a pretty much random mix of those include files used throughout the tree. In order to detangle these headers and replace the implicit includes with struct declarations, users need to explicitly include the correct includes. Signed-off-by: Rob Herring <robh(a)kernel.org> --- drivers/soundwire/qcom.c | 1 - 1 file changed, 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 7970fdb27ba0..d178a0dc0918 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -10,7 +10,6 @@ #include <linux/debugfs.h> #include <linux/of.h> #include <linux/of_irq.h> -#include <linux/of_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/reset.h> -- 2.40.1

2 1

[PATCH v2] ALSA: seq: Fix memory leak at error path in snd_seq_create_port()
by Takashi Iwai 17 Jul '23

17 Jul '23

We forgot to release a newly allocated item at the error path in snd_seq_create_port(). This patch fixes it. Fixes: 7c3f0d3d3a11 ("ALSA: seq: Check the conflicting port at port creation") Reported-by: syzbot+cf8e7fa4eeec59b3d485(a)syzkaller.appspotmail.com Closes: https://lore.kernel.org/r/00000000000098ed3a0600965f89@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: only patch meta data correction sound/core/seq/seq_ports.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 9b80f8275026..f3f14ff0f80f 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -149,6 +149,7 @@ int snd_seq_create_port(struct snd_seq_client *client, int port, write_lock_irq(&client->ports_lock); list_for_each_entry(p, &client->ports_list_head, list) { if (p->addr.port == port) { + kfree(new_port); num = -EBUSY; goto unlock; } -- 2.35.3

1 0

[GIT PULL] ASoC fixes for v6.5-rc1
by Mark Brown 17 Jul '23

17 Jul '23

The following changes since commit 2d0cad0473bd1ffbc5842be0b9f2546265acb011: ASoC: core: Always store of_node when getting DAI link component (2023-06-23 22:09:00 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc1 for you to fetch changes up to e8bf1741c14eb8e4a4e1364d45aeeab66660ab9b: MAINTAINERS: Redo addition of ssm3515 to APPLE SOUND (2023-07-15 13:07:32 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.5 A lot of fixes here for the Qualcomm CODEC drivers, there was quite a bit of fragility with the SoundWire probe due to the combined DT and hotplug approach that the bus has which Johan Hovold fixed along with a bunch of other issues that came up in the process. Srivinvas Kandagatla also fixed some separate issues that have been lurking for a while in the Qualcomm AP side, and there's a good set of AMD fixes from Vijendar Mukunda too. ---------------------------------------------------------------- Derek Fang (1): ASoC: rt5640: Fix the issue of speaker noise Fabio Estevam (1): ASoC: fsl_sai: Revert "ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode" Geert Uytterhoeven (1): ASoC: codecs: SND_SOC_WCD934X should select REGMAP_IRQ Johan Hovold (15): ASoC: codecs: wcd938x: fix codec initialisation race ASoC: codecs: wcd938x: fix mbhc impedance loglevel ASoC: codecs: wcd938x: drop inline keywords ASoC: codecs: wcd938x: use dev_printk() for impedance logging ASoC: codecs: wcd934x: demote impedance printk ASoC: codecs: wcd934x: drop inline keywords ASoC: codecs: wcd938x: fix soundwire initialisation race ASoC: codecs: wcd938x: fix missing mbhc init error handling ASoC: qdsp6: audioreach: fix topology probe deferral ASoC: codecs: wcd938x: fix missing clsh ctrl error handling ASoC: codecs: wcd938x: fix resource leaks on component remove ASoC: codecs: wcd934x: fix resource leaks on component remove ASoC: codecs: wcd-mbhc-v2: fix resource leaks on component remove ASoC: topology: suppress probe deferral errors ASoC: core: suppress probe deferral errors Mario Limonciello (1): ASoC: amd: ps: Fix extraneous error messages Mark Brown (2): ASoC: codecs: wcd938x/wcd934x: loglevel fix and ASoC/soundwire/qdsp6/wcd: fix leaks and probe deferral Martin Povišer (1): MAINTAINERS: Redo addition of ssm3515 to APPLE SOUND Mastan Katragadda (1): ASoC: SOF: amd: add revision check for sending sha dma completion command Matthias Reichl (1): ASoC: hdmi-codec: fix channel info for compressed formats Nathan Chancellor (1): ASoC: cs35l45: Select REGMAP_IRQ Rob Herring (1): ASoC: dt-bindings: audio-graph-card2: Drop incomplete example Rohit kumar (1): ASoC: dt-bindings: Update maintainer email id Sameer Pujar (1): ASoC: rt5640: Fix sleep in atomic context Sheetal (2): ASoC: tegra: Fix AMX byte map ASoC: tegra: Fix ADX byte map Shuming Fan (1): ASoC: rt5645: check return value after reading device id Srinivas Kandagatla (4): ASoC: qdsp6: q6apm: use dai link pcm id as pcm device number ASoC: qcom: q6afe-dai: fix Display Port Playback stream name ASoC: codecs: wcd938x: fix dB range for HPHL and HPHR ASoC: qcom: q6apm: do not close GPR port before closing graph Vijendar Mukunda (6): ASoC: amd: ps: add comments for DMA irq bits mapping ASoC: amd: ps: add fix for dma irq mask for rx streams for SDW0 instance ASoC: amd: ps: fix for position register set for AUDIO0 RX stream ASoC: amd: ps: add comments for DMA register mapping ASoC: amd: ps: fix byte count return value for invalid SoundWire manager instance ASoC: amd: acp: fix for invalid dai id handling in acp_get_byte_count() .../bindings/sound/audio-graph-card2.yaml | 20 +---- .../bindings/sound/google,sc7180-trogdor.yaml | 2 +- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 2 +- MAINTAINERS | 2 + sound/soc/amd/acp/amd.h | 7 +- sound/soc/amd/ps/acp63.h | 22 ++++- sound/soc/amd/ps/pci-ps.c | 4 +- sound/soc/amd/ps/ps-sdw-dma.c | 16 +++- sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/hdmi-codec.c | 36 +++++--- sound/soc/codecs/rt5640.c | 13 +-- sound/soc/codecs/rt5645.c | 6 +- sound/soc/codecs/wcd-mbhc-v2.c | 57 +++++++++---- sound/soc/codecs/wcd934x.c | 20 ++++- sound/soc/codecs/wcd938x.c | 99 ++++++++++++++++++---- sound/soc/fsl/fsl_sai.c | 6 -- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 1 + sound/soc/qcom/qdsp6/q6apm.c | 7 +- sound/soc/qcom/qdsp6/topology.c | 4 +- sound/soc/soc-core.c | 6 +- sound/soc/soc-topology.c | 10 ++- sound/soc/sof/amd/acp.c | 10 ++- sound/soc/tegra/tegra210_adx.c | 34 +++++--- sound/soc/tegra/tegra210_amx.c | 40 +++++---- 25 files changed, 294 insertions(+), 134 deletions(-)

2 1

[PATCH] ALSA: seq: Fix memory leak at error path in snd_seq_create_port()
by Takashi Iwai 17 Jul '23

17 Jul '23

We forgot to release a newly allocated item at the error path in snd_seq_create_port(). This patch fixes it. Reported-by: syzbot+cf8e7fa4eeec59b3d485(a)syzkaller.appspotmail.com Closes: https://lore.kernel.org/r/00000000000098ed3a0600965f89@google.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_ports.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 9b80f8275026..f3f14ff0f80f 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -149,6 +149,7 @@ int snd_seq_create_port(struct snd_seq_client *client, int port, write_lock_irq(&client->ports_lock); list_for_each_entry(p, &client->ports_list_head, list) { if (p->addr.port == port) { + kfree(new_port); num = -EBUSY; goto unlock; } -- 2.35.3

1 0

Pass through encoded audio in the AC3 profile
by GitHub issues - opened 16 Jul '23

16 Jul '23

alsa-project/alsa-plugins issue #53 was opened from AndrewAmmerlaan: Is there any technical reason why it is not possible to `pactl set-sink-formats X "pcm dts-iec61937 ac3-iec61937"` when the `Digital Surround 5.1 IEC958/AC3 Output` profile is selected? It would be convenient to not have to switch profiles when watching a movie with audio that is already encoded AC3. I noticed the `dcaenc` plug-in for DTS encoding does the same thing. So maybe there is a good reason for this I'm not seeing. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/53 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ASoC: dt-bindings: audio-graph-card2: Drop incomplete example
by Rob Herring 16 Jul '23

16 Jul '23

The example in audio-graph-card2 binding is incomplete, uses undocumented compatibles strings, and doesn't follow typical .dts formatting. Rather than try to fix with what would probably be a lengthy example, just drop the example. Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/audio-graph-card2.yaml | 20 +------------------ 1 file changed, 1 insertion(+), 19 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml index 3de7b36829da..d3ce4de449d5 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml @@ -39,22 +39,4 @@ required: additionalProperties: false -examples: - - | - sound { - compatible = "audio-graph-card2"; - - links = <&cpu_port>; - }; - - cpu { - compatible = "cpu-driver"; - - cpu_port: port { cpu_ep: endpoint { remote-endpoint = <&codec_ep>; }; }; - }; - - codec { - compatible = "codec-driver"; - - port { codec_ep: endpoint { remote-endpoint = <&cpu_ep>; }; }; - }; +... -- 2.40.1

2 2

[PATCH -next] ASoC: amd: acp: clean up some inconsistent indentings
by Yang Li 16 Jul '23

16 Jul '23

sound/soc/amd/acp/acp-rembrandt.c:283 rmb_pcm_resume() warn: inconsistent indenting Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Closes: https://bugzilla.openanolis.cn/show_bug.cgi?id=5863 Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/amd/acp/acp-rembrandt.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-rembrandt.c b/sound/soc/amd/acp/acp-rembrandt.c index 89314d95ec2b..21e67ed956d1 100644 --- a/sound/soc/amd/acp/acp-rembrandt.c +++ b/sound/soc/amd/acp/acp-rembrandt.c @@ -280,8 +280,8 @@ static int __maybe_unused rmb_pcm_resume(struct device *dev) } } } - spin_unlock(&adata->acp_lock); - return 0; + spin_unlock(&adata->acp_lock); + return 0; } static const struct dev_pm_ops rmb_dma_pm_ops = { -- 2.20.1.7.g153144c

2 2

[PATCH] ALSA: Explicitly include correct DT includes
by Rob Herring 16 Jul '23

16 Jul '23

The DT of_device.h and of_platform.h date back to the separate of_platform_bus_type before it as merged into the regular platform bus. As part of that merge prepping Arm DT support 13 years ago, they "temporarily" include each other. They also include platform_device.h and of.h. As a result, there's a pretty much random mix of those include files used throughout the tree. In order to detangle these headers and replace the implicit includes with struct declarations, users need to explicitly include the correct includes. Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/atmel/ac97c.c | 3 +-- sound/drivers/serial-generic.c | 2 +- sound/pci/hda/hda_tegra.c | 3 ++- sound/ppc/awacs.c | 1 + sound/ppc/burgundy.c | 1 + sound/sparc/amd7930.c | 3 +-- sound/sparc/cs4231.c | 2 +- sound/sparc/dbri.c | 2 +- 8 files changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index c8912b8a1dc5..402b5f66dcc3 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -12,13 +12,12 @@ #include <linux/gpio/consumer.h> #include <linux/init.h> #include <linux/interrupt.h> +#include <linux/mod_devicetable.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/mutex.h> #include <linux/types.h> #include <linux/io.h> -#include <linux/of.h> -#include <linux/of_device.h> #include <sound/core.h> #include <sound/initval.h> diff --git a/sound/drivers/serial-generic.c b/sound/drivers/serial-generic.c index e1f864dc7939..b0262541802a 100644 --- a/sound/drivers/serial-generic.c +++ b/sound/drivers/serial-generic.c @@ -16,7 +16,7 @@ #include <linux/io.h> #include <linux/ioport.h> #include <linux/module.h> -#include <linux/of_device.h> +#include <linux/of.h> #include <linux/serdev.h> #include <linux/serial_reg.h> #include <linux/slab.h> diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 9d0ab043880b..39fa036616ce 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -16,7 +16,8 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/mutex.h> -#include <linux/of_device.h> +#include <linux/of.h> +#include <linux/platform_device.h> #include <linux/reset.h> #include <linux/slab.h> #include <linux/time.h> diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 53d558b2806c..659866cfe3b4 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -11,6 +11,7 @@ #include <asm/nvram.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/of.h> #include <linux/slab.h> #include <sound/core.h> #include "pmac.h" diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 4fb990ab2ceb..400a886562b1 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -9,6 +9,7 @@ #include <linux/io.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/of.h> #include <sound/core.h> #include "pmac.h" #include "burgundy.h" diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index c434b69a83f1..0fea04acc3ea 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -37,7 +37,7 @@ #include <linux/interrupt.h> #include <linux/moduleparam.h> #include <linux/of.h> -#include <linux/of_device.h> +#include <linux/platform_device.h> #include <linux/io.h> #include <sound/core.h> @@ -47,7 +47,6 @@ #include <sound/initval.h> #include <asm/irq.h> -#include <asm/prom.h> static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 31bac355ec4d..c2ad3fa2f25a 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -18,7 +18,7 @@ #include <linux/irq.h> #include <linux/io.h> #include <linux/of.h> -#include <linux/of_device.h> +#include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 376aed136a45..050e98f32d36 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -69,7 +69,7 @@ #include <sound/initval.h> #include <linux/of.h> -#include <linux/of_device.h> +#include <linux/platform_device.h> #include <linux/atomic.h> #include <linux/module.h> -- 2.40.1

2 1

speaker-test: inconsistent behaviour of --buffer option.
by GitHub issues - edited 16 Jul '23

16 Jul '23

alsa-project/alsa-utils issue #224 was edited from AndrewAmmerlaan: Without specifying the `--buffer` option `speaker-test` returns this: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Using max buffer size 1048576 Periods = 4 was set period_size = 262144 was set buffer_size = 1048576 0 - Front Left 1 - Front Right Write error: -77,File descriptor in bad state xrun_recovery failed: -77,File descriptor in bad state Transfer failed: File descriptor in bad state ``` Note that it says `Using max buffer size 1048576`. But now if we try to explicitly specify this maximum buffer size: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 --buffer 1048576 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Requested buffer time 1000000 us Periods = 4 was set period_size = 12000 was set buffer_size = 48000 0 - Front Left 1 - Front Right Time per period = 2.251287 ``` Note that now we get `Requested buffer time 1000000 us` which is less then what we actually specified. In fact this will happen with any `--buffer` parameter larger then `1000000`: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 --buffer 1000001 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Requested buffer time 1000000 us Periods = 4 was set period_size = 12000 was set buffer_size = 48000 0 - Front Left 1 - Front Right Time per period = 2.251042 ``` Leading me to believe that despite what `speaker-test` claims, `1000000` is actually the maximum, not `1048576`. Discovered while debugging https://gitlab.freedesktop.org/drm/intel/-/issues/8462 This behaviour of `speaker-test` is confusing though and feels to me like a separate issue. Issue URL : https://github.com/alsa-project/alsa-utils/issues/224 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

speaker-test: inconsistent behaviour of --buffer option.
by GitHub issues - opened 16 Jul '23

16 Jul '23

alsa-project/alsa-utils issue #224 was opened from AndrewAmmerlaan: Without specifying the `--buffer` option `speaker-test returns this: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Using max buffer size 1048576 Periods = 4 was set period_size = 262144 was set buffer_size = 1048576 0 - Front Left 1 - Front Right Write error: -77,File descriptor in bad state xrun_recovery failed: -77,File descriptor in bad state Transfer failed: File descriptor in bad state ``` Note that it says `Using max buffer size 1048576`. But now if we try to explicitly specify this maximum buffer size: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 --buffer 1048576 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Requested buffer time 1000000 us Periods = 4 was set period_size = 12000 was set buffer_size = 48000 0 - Front Left 1 - Front Right Time per period = 2.251287 ``` Note that now we get `Requested buffer time 1000000 us` which is less then what we actually specified. In fact this will happen with any `--buffer` parameter larger then `1000000`: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 --buffer 1000001 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Requested buffer time 1000000 us Periods = 4 was set period_size = 12000 was set buffer_size = 48000 0 - Front Left 1 - Front Right Time per period = 2.251042 ``` Leading me to believe that despite what `speaker-test` claims, `1000000` is actually the maximum, not `1048576`. Discovered while debugging https://gitlab.freedesktop.org/drm/intel/-/issues/8462 This behaviour of `speaker-test` is confusing though and feels to me like a separate issue. Issue URL : https://github.com/alsa-project/alsa-utils/issues/224 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 1/2] ALSA: hda: cs35l41: Add fixups for machines without a valid ACPI _DSD
by David Xu 15 Jul '23

15 Jul '23

As the comments added in commit 4d4c4bff4f8ed79d95e05 ("ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties"), CSC3551 requires a valid _DSD to work and the current implementation just fails when no _DSD can be found for CSC3551. However it is a fact that many OEMs hardcoded the configurations needed by CSC3551 into their proprietary software for various 2022 and later laptop models, and this makes the Linux installations on these models cannot make any speaker sound. Meanwhile, at this point of time, we see no hope that these OEMs would ever fix this problem via a BIOS update. So to fix this bothering problem it might be worthwhile to add some model-specific fixups to apply some proper configurations to the cs35l41. To address the above problem, a configuration fixup function apply_cs35l41_fixup_cfg that would be called in cs35l41_no_acpi_dsd, along with a fixup table cs35l41_fixup_cfgtbl which is a array of fixup entry struct cs35l41_fixup_cfg are introduced. Each fixup entry records the ACPI _SUB(vender and device ID) to be matched, and a configuration to be applied to each of the cs35l41 device in CSC3551. More specifically for the design of this fixup mechanism, the maximum number of cs35l41 configurations inside a fixup entry is defined as a macro CS35L41_FIXUP_CFG_MAX_DEVICES, and the actual number of cs35l41 devices in a CSC3551 system is recorded in the num_device field in the fixup entry. The apply_cs35l41_fixup_cfg function is responsible for finding and applying a fixup for a specific model according to the model's ACPI _SUB. If no valid fixup can be applied, the fixup function fails and returns to the normal cs35l41_no_acpi_dsd execution flow. This patch now contains only several fixups for three Lenovo laptop models, namely 16IAH7, 16IAX7, and 16ARHA7 and this fixup mechanism has been verified to work on 16IAH7. As far as is known, several other laptop models from ASUS and HP also suffer from this no valid _DSD problem and could have it addressed with this fixup mechanism when proper fixup entries are inserted. Signed-off-by: David Xu <xuwd1(a)hotmail.com> --- sound/pci/hda/cs35l41_hda.c | 160 ++++++++++++++++++++++++++++++++++++ 1 file changed, 160 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index ce5faa620517..d957458dd4e6 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1211,6 +1211,159 @@ static int cs35l41_get_speaker_id(struct device *dev, int amp_index, return speaker_id; } +#define CS35L41_FIXUP_CFG_MAX_DEVICES 4 + +struct cs35l41_fixup_cfg { + unsigned short vender; + unsigned short device; + unsigned int num_device; /* The num of cs35l41 instances */ + /* cs35l41 instance ids, can be i2c index or spi index */ + int ids[CS35L41_FIXUP_CFG_MAX_DEVICES]; + unsigned int reset_gpio_idx[CS35L41_FIXUP_CFG_MAX_DEVICES]; + enum gpiod_flags reset_gpio_flags[CS35L41_FIXUP_CFG_MAX_DEVICES]; + int spkid_gpio_idx[CS35L41_FIXUP_CFG_MAX_DEVICES]; + unsigned int spk_pos[CS35L41_FIXUP_CFG_MAX_DEVICES]; + enum cs35l41_hda_gpio_function gpio1_func[CS35L41_FIXUP_CFG_MAX_DEVICES]; + enum cs35l41_hda_gpio_function gpio2_func[CS35L41_FIXUP_CFG_MAX_DEVICES]; + enum cs35l41_boost_type bst_type[CS35L41_FIXUP_CFG_MAX_DEVICES]; +}; + +static const struct cs35l41_fixup_cfg cs35l41_fixup_cfgtbl[] = { + { // Lenovo Legion Slim 7i 16IAH7 + .vender = 0x17aa, + .device = 0x386e, + .num_device = 2, + .ids = {0x40, 0x41}, + .reset_gpio_idx = {0, 0}, + .reset_gpio_flags = {GPIOD_OUT_LOW, GPIOD_OUT_LOW}, + .spkid_gpio_idx = {1, 1}, + .spk_pos = {0, 1}, + .gpio1_func = {CS35l41_VSPK_SWITCH, CS35l41_VSPK_SWITCH}, + .gpio2_func = {CS35L41_INTERRUPT, CS35L41_NOT_USED}, + .bst_type = {CS35L41_EXT_BOOST, CS35L41_EXT_BOOST} + }, + { // Lenovo Legion Slim 7i 16IAH7 type2 + .vender = 0x17aa, + .device = 0x3803, + .num_device = 2, + .ids = {0x40, 0x41}, + .reset_gpio_idx = {0, 0}, + .reset_gpio_flags = {GPIOD_OUT_LOW, GPIOD_OUT_LOW}, + .spkid_gpio_idx = {1, 1}, + .spk_pos = {0, 1}, + .gpio1_func = {CS35l41_VSPK_SWITCH, CS35l41_VSPK_SWITCH}, + .gpio2_func = {CS35L41_INTERRUPT, CS35L41_NOT_USED}, + .bst_type = {CS35L41_EXT_BOOST, CS35L41_EXT_BOOST} + }, + { // Lenovo Legion 7i 16IAX7 + .vender = 0x17aa, + .device = 0x3874, + .num_device = 2, + .ids = {0x40, 0x41}, + .reset_gpio_idx = {0, 0}, + .reset_gpio_flags = {GPIOD_OUT_LOW, GPIOD_OUT_LOW}, + .spkid_gpio_idx = {1, 1}, + .spk_pos = {0, 1}, + .gpio1_func = {CS35l41_VSPK_SWITCH, CS35l41_VSPK_SWITCH}, + .gpio2_func = {CS35L41_INTERRUPT, CS35L41_INTERRUPT}, + .bst_type = {CS35L41_EXT_BOOST, CS35L41_EXT_BOOST} + }, + { // Lenovo Legion 7i 16IAX7 type 2 + .vender = 0x17aa, + .device = 0x386f, + .num_device = 2, + .ids = {0x40, 0x41}, + .reset_gpio_idx = {0, 0}, + .reset_gpio_flags = {GPIOD_OUT_LOW, GPIOD_OUT_LOW}, + .spkid_gpio_idx = {1, 1}, + .spk_pos = {0, 1}, + .gpio1_func = {CS35l41_VSPK_SWITCH, CS35l41_VSPK_SWITCH}, + .gpio2_func = {CS35L41_INTERRUPT, CS35L41_INTERRUPT}, + .bst_type = {CS35L41_EXT_BOOST, CS35L41_EXT_BOOST} + }, + { // Lenovo Legion Slim 7 16ARHA7 + .vender = 0x17aa, + .device = 0x3877, + .num_device = 2, + .ids = {0x40, 0x41}, + .reset_gpio_idx = {0, 0}, + .reset_gpio_flags = {GPIOD_OUT_LOW, GPIOD_OUT_LOW}, + .spkid_gpio_idx = {1, 1}, + .spk_pos = {0, 1}, + .gpio1_func = {CS35l41_VSPK_SWITCH, CS35l41_VSPK_SWITCH}, + .gpio2_func = {CS35L41_INTERRUPT, CS35L41_INTERRUPT}, + .bst_type = {CS35L41_EXT_BOOST, CS35L41_EXT_BOOST} + }, + {} // terminator +}; + +static inline int cs35l41_fixup_get_index(const struct cs35l41_fixup_cfg *fixup, int cs35l41_addr) +{ + int i; + + for (i = 0; i < fixup->num_device; i++) { + if (fixup->ids[i] == cs35l41_addr) + return i; + } + return -ENODEV; +} + +static int apply_cs35l41_fixup_cfg(struct cs35l41_hda *cs35l41, + struct device *physdev, + int cs35l41_addr, + const struct cs35l41_fixup_cfg *fixup_tbl) +{ + const char *ssid; + unsigned int vendid; + unsigned int devid; + const struct cs35l41_fixup_cfg *cur_fixup; + struct cs35l41_hw_cfg *hw_cfg; + int cs35l41_index; + int ret; + int i; + + ssid = cs35l41->acpi_subsystem_id; + ret = sscanf(ssid, "%04x%04x", &vendid, &devid); + if (ret != 2) + return -EINVAL; + + hw_cfg = &cs35l41->hw_cfg; + for (cur_fixup = fixup_tbl; cur_fixup->vender; cur_fixup++) { + if (cur_fixup->vender == vendid && cur_fixup->device == devid) { + cs35l41_index = cs35l41_fixup_get_index(cur_fixup, cs35l41_addr); + if (cs35l41_index == -ENODEV) + return -ENODEV; + cs35l41->index = cs35l41_index; + cs35l41->reset_gpio = gpiod_get_index( + physdev, + NULL, + cur_fixup->reset_gpio_idx[cs35l41_index], + cur_fixup->reset_gpio_flags[cs35l41_index] + ); + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, + cs35l41_index, + cur_fixup->num_device, + cur_fixup->spkid_gpio_idx[cs35l41_index] + ); + hw_cfg->spk_pos = cur_fixup->spk_pos[cs35l41_index]; + cs35l41->channel_index = 0; + for (i = 0; i < cs35l41->index; i++) + if (cur_fixup->spk_pos[i] == hw_cfg->spk_pos) + cs35l41->channel_index++; + + hw_cfg->gpio1.func = cur_fixup->gpio1_func[cs35l41_index]; + hw_cfg->gpio1.valid = true; + hw_cfg->gpio2.func = cur_fixup->gpio2_func[cs35l41_index]; + hw_cfg->gpio2.valid = true; + hw_cfg->bst_type = cur_fixup->bst_type[cs35l41_index]; + dev_dbg(physdev, "Fixup applied.\n"); + break; + } + } + return 0; + +} + /* * Device CLSA010(0/1) doesn't have _DSD so a gpiod_get by the label reset won't work. * And devices created by serial-multi-instantiate don't have their device struct @@ -1221,6 +1374,7 @@ static int cs35l41_get_speaker_id(struct device *dev, int amp_index, static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physdev, int id, const char *hid) { + int ret; struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; /* check I2C address to assign the index */ @@ -1243,7 +1397,13 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd /* * Note: CLSA010(0/1) are special cases which use a slightly different design. * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. + * However many OEMs hardcoded the configurations into their proprietary software + * thus leaving our Linux installation with no speaker sound at all while we see + * no hope those OEMs would fix it. So we apply a ssid specific fixup to fix it. */ + if (apply_cs35l41_fixup_cfg(cs35l41, physdev, id, cs35l41_fixup_cfgtbl) == 0) + return 0; + dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.\n", hid); hw_cfg->valid = false; hw_cfg->gpio1.valid = false; -- 2.41.0

2 3

RE: [EXTERNAL] {standard input}:866: Error: value -148 out of range
by Ding, Shenghao 15 Jul '23

15 Jul '23

> > From：kernel test robot <mailto:lkp@intel.com> > To：Shenghao Ding <mailto:13916275206@139.com> > Date：2023-07-10 04:59:51 > Subject：{standard input}:866: Error: value -148 out of range > > tree: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git mast > er > head: c192ac7357683f78c2e6d6e75adfcc29deb8c4ae > commit: ef3bcde75d06d65f78ba38a30d5a87fb83a5cdae ASoC: tas2781: Add > tas2781 driver > date: 3 weeks ago > config: m68k-randconfig-r015-20230710 (https://download.01.org/0day- > ci/archive/20230710/202307100413.lPlgpaPd-lkp(a)intel.com/config) > compiler: m68k-linux-gcc (GCC) 12.3.0 > reproduce: (https://download.01.org/0day- > ci/archive/20230710/202307100413.lPlgpaPd-lkp(a)intel.com/reproduce) > > If you fix the issue in a separate patch/commit (i.e. not just a new version of > the same patch/commit), kindly add following tags > | Reported-by: kernel test robot <mailto:lkp@intel.com> > | Closes: > |https://lore.kernel.org/oe-kbuild-all/202307100413.lPlgpaPd-lkp@intel.c > |om/ > > All errors (new ones prefixed by >>): > > {standard input}: Assembler messages: > >> {standard input}:866: Error: value -148 out of range > >> {standard input}:866: Error: value of ffffff6c too large for field of > >>1 byte at 0000043b Would you be so kind and give me some advice on how to I fix this issue? > > -- > 0-DAY CI Kernel Test Service > https://github.com/intel/lkp-tests/wiki

2 1

[PATCH v1 1/5] ASoC: codecs: ES8326: Add es8326_mute function
by Zhu Ning 15 Jul '23

15 Jul '23

The internal analog power and hp Vref of es8326 should always be on to reduce pop noise. The HP_VOL and HP_CAL are moved to es8326_mute function so they are turned on at last and turned off at first. Also, the calibration should be done manually once during start-up to reduce DC offset on headphone. Signed-off-by: Zhu Ning <zhuning0077(a)gmail.com> --- sound/soc/codecs/es8326.c | 75 ++++++++++++++++----------------------- sound/soc/codecs/es8326.h | 9 +++-- 2 files changed, 37 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index ed348bbf5f49..191579d9552c 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -38,6 +38,9 @@ struct es8326_priv { u8 interrupt_clk; bool jd_inverted; unsigned int sysclk; + + bool calibrated; + int version; }; static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); @@ -121,33 +124,12 @@ static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { /* Analog Power Supply*/ SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN, 0, 1), SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN, 1, 1), - SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN, 7, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN, 6, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN, 5, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN, 4, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN, 3, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX, 3, 0, NULL, 0), - /* Headphone Charge Pump and Output */ - SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL, 7, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL, 3, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER, - 3, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER, - 2, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER, - 1, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER, - 0, 1, NULL, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL, - ES8326_HPOR_SHIFT, 7, 7, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL, - 0, 7, 7, 0), - SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), }; @@ -166,34 +148,12 @@ static const struct snd_soc_dapm_route es8326_dapm_routes[] = { {"I2S OUT", NULL, "ADC L"}, {"I2S OUT", NULL, "ADC R"}, - {"I2S OUT", NULL, "Analog Power"}, - {"I2S OUT", NULL, "ADC Vref"}, - {"I2S OUT", NULL, "Vref Power"}, - {"I2S OUT", NULL, "IBias Power"}, - {"I2S IN", NULL, "Analog Power"}, - {"I2S IN", NULL, "DAC Vref"}, - {"I2S IN", NULL, "Vref Power"}, - {"I2S IN", NULL, "IBias Power"}, - {"Right DAC", NULL, "I2S IN"}, {"Left DAC", NULL, "I2S IN"}, {"LHPMIX", NULL, "Left DAC"}, {"RHPMIX", NULL, "Right DAC"}, - {"HPOR", NULL, "HPOR Cal"}, - {"HPOL", NULL, "HPOL Cal"}, - {"HPOR", NULL, "HPOR Supply"}, - {"HPOL", NULL, "HPOL Supply"}, - {"HPOL", NULL, "Headphone Charge Pump"}, - {"HPOR", NULL, "Headphone Charge Pump"}, - {"HPOL", NULL, "Headphone Driver Bias"}, - {"HPOR", NULL, "Headphone Driver Bias"}, - {"HPOL", NULL, "Headphone LDO"}, - {"HPOR", NULL, "Headphone LDO"}, - {"HPOL", NULL, "Headphone Reference"}, - {"HPOR", NULL, "Headphone Reference"}, - {"HPOL", NULL, "LHPMIX"}, {"HPOR", NULL, "RHPMIX"}, }; @@ -419,6 +379,31 @@ static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } +static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) +{ + struct snd_soc_component *component = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + if (mute) { + regmap_write(es8326->regmap, ES8326_HP_CAL, ES8326_HP_OFF); + regmap_update_bits(es8326->regmap, ES8326_DAC_MUTE, + ES8326_MUTE_MASK, ES8326_MUTE); + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xf0); + } else { + if (!es8326->calibrated) { + regmap_write(es8326->regmap, ES8326_HP_CAL, ES8326_HP_FORCE_CAL); + msleep(30); + es8326->calibrated = true; + } + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xa0); + regmap_write(es8326->regmap, ES8326_HP_VOL, 0x00); + regmap_write(es8326->regmap, ES8326_HP_CAL, ES8326_HP_ON); + regmap_update_bits(es8326->regmap, ES8326_DAC_MUTE, + ES8326_MUTE_MASK, ~(ES8326_MUTE)); + } + return 0; +} + static int es8326_set_bias_level(struct snd_soc_component *codec, enum snd_soc_bias_level level) { @@ -469,6 +454,8 @@ static const struct snd_soc_dai_ops es8326_ops = { .hw_params = es8326_pcm_hw_params, .set_fmt = es8326_set_dai_fmt, .set_sysclk = es8326_set_dai_sysclk, + .mute_stream = es8326_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver es8326_dai = { @@ -691,7 +678,7 @@ static int es8326_suspend(struct snd_soc_component *component) cancel_delayed_work_sync(&es8326->jack_detect_work); es8326_disable_micbias(component); - + es8326->calibrated = false; regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); regcache_cache_only(es8326->regmap, true); regcache_mark_dirty(es8326->regmap); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h index 8e5ffe5ee10d..3f8f7da58062 100644 --- a/sound/soc/codecs/es8326.h +++ b/sound/soc/codecs/es8326.h @@ -9,8 +9,6 @@ #ifndef _ES8326_H #define _ES8326_H -#define CONFIG_HHTECH_MINIPMP 1 - /* ES8326 register space */ #define ES8326_RESET 0x00 #define ES8326_CLK_CTL 0x01 @@ -94,6 +92,8 @@ #define ES8326_PWRUP_SEQ_EN (1 << 5) #define ES8326_CODEC_RESET (0x0f << 0) #define ES8326_CSM_OFF (0 << 7) +#define ES8326_MUTE_MASK (3 << 0) +#define ES8326_MUTE (3 << 0) /* ES8326_CLK_CTL */ #define ES8326_CLK_ON (0x7f << 0) @@ -122,7 +122,9 @@ #define ES8326_MIC2_SEL (1 << 5) /* ES8326_HP_CAL */ -#define ES8326_HPOR_SHIFT 4 +#define ES8326_HP_OFF 0 +#define ES8326_HP_FORCE_CAL ((1 << 7) | (1 << 3)) +#define ES8326_HP_ON ((7 << 4) | (7 << 0)) /* ES8326_ADC1_SRC */ #define ES8326_ADC1_SHIFT 0 @@ -180,3 +182,4 @@ #define ES8326_VERSION_B (1 << 0) #endif + -- 2.34.1

2 7

[PATCH] ASoC: rt5640: Fix the issue of speaker noise
by derek.fang＠realtek.com 14 Jul '23

14 Jul '23

From: Derek Fang <derek.fang(a)realtek.com> Remove the class-D internal register setting during initialization to be compatible with most speaker designs to avoid noise. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5640.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e24ed754ac56..eceed8209787 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -53,7 +53,6 @@ static const struct reg_sequence init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, - {RT5640_PR_BASE + 0x20, 0x6110}, {RT5640_PR_BASE + 0x21, 0xe0e0}, {RT5640_PR_BASE + 0x23, 0x1804}, }; -- 2.17.1

2 1

[PATCH] irqchip: Explicitly include correct DT includes
by Rob Herring 14 Jul '23

14 Jul '23

The DT of_device.h and of_platform.h date back to the separate of_platform_bus_type before it as merged into the regular platform bus. As part of that merge prepping Arm DT support 13 years ago, they "temporarily" include each other. They also include platform_device.h and of.h. As a result, there's a pretty much random mix of those include files used throughout the tree. In order to detangle these headers and replace the implicit includes with struct declarations, users need to explicitly include the correct includes. Signed-off-by: Rob Herring <robh(a)kernel.org> --- drivers/irqchip/irq-bcm6345-l1.c | 1 - drivers/irqchip/irq-bcm7038-l1.c | 1 - drivers/irqchip/irq-brcmstb-l2.c | 1 - drivers/irqchip/irq-gic-pm.c | 2 +- drivers/irqchip/irq-gic-v3-its-fsl-mc-msi.c | 2 -- drivers/irqchip/irq-imx-intmux.c | 3 ++- drivers/irqchip/irq-imx-irqsteer.c | 3 ++- drivers/irqchip/irq-keystone.c | 2 +- drivers/irqchip/irq-loongson-htvec.c | 1 - drivers/irqchip/irq-loongson-pch-pic.c | 2 +- drivers/irqchip/irq-madera.c | 4 +--- drivers/irqchip/irq-pruss-intc.c | 2 +- drivers/irqchip/irq-qcom-mpm.c | 2 +- drivers/irqchip/irq-renesas-intc-irqpin.c | 1 - drivers/irqchip/irq-st.c | 2 +- drivers/irqchip/irq-stm32-exti.c | 3 ++- drivers/irqchip/irq-sunxi-nmi.c | 1 - drivers/irqchip/irq-tb10x.c | 1 - drivers/irqchip/irq-ti-sci-inta.c | 4 ++-- drivers/irqchip/irq-ti-sci-intr.c | 4 ++-- drivers/irqchip/irq-uniphier-aidet.c | 1 - drivers/irqchip/irqchip.c | 2 +- drivers/irqchip/qcom-pdc.c | 1 - 23 files changed, 18 insertions(+), 28 deletions(-) diff --git a/drivers/irqchip/irq-bcm6345-l1.c b/drivers/irqchip/irq-bcm6345-l1.c index fa113cb2529a..b521897277a8 100644 --- a/drivers/irqchip/irq-bcm6345-l1.c +++ b/drivers/irqchip/irq-bcm6345-l1.c @@ -60,7 +60,6 @@ #include <linux/of.h> #include <linux/of_irq.h> #include <linux/of_address.h> -#include <linux/of_platform.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/smp.h> diff --git a/drivers/irqchip/irq-bcm7038-l1.c b/drivers/irqchip/irq-bcm7038-l1.c index a62b96237b82..24ca1d656adc 100644 --- a/drivers/irqchip/irq-bcm7038-l1.c +++ b/drivers/irqchip/irq-bcm7038-l1.c @@ -20,7 +20,6 @@ #include <linux/of.h> #include <linux/of_irq.h> #include <linux/of_address.h> -#include <linux/of_platform.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/smp.h> diff --git a/drivers/irqchip/irq-brcmstb-l2.c b/drivers/irqchip/irq-brcmstb-l2.c index 091b0fe7e324..5559c943f03f 100644 --- a/drivers/irqchip/irq-brcmstb-l2.c +++ b/drivers/irqchip/irq-brcmstb-l2.c @@ -15,7 +15,6 @@ #include <linux/of.h> #include <linux/of_irq.h> #include <linux/of_address.h> -#include <linux/of_platform.h> #include <linux/interrupt.h> #include <linux/irq.h> #include <linux/io.h> diff --git a/drivers/irqchip/irq-gic-pm.c b/drivers/irqchip/irq-gic-pm.c index 3989d16f997b..a275a8071a25 100644 --- a/drivers/irqchip/irq-gic-pm.c +++ b/drivers/irqchip/irq-gic-pm.c @@ -4,7 +4,7 @@ */ #include <linux/module.h> #include <linux/clk.h> -#include <linux/of_device.h> +#include <linux/of.h> #include <linux/of_irq.h> #include <linux/irqchip/arm-gic.h> #include <linux/platform_device.h> diff --git a/drivers/irqchip/irq-gic-v3-its-fsl-mc-msi.c b/drivers/irqchip/irq-gic-v3-its-fsl-mc-msi.c index 634263dfd7b5..8e87fc35f8aa 100644 --- a/drivers/irqchip/irq-gic-v3-its-fsl-mc-msi.c +++ b/drivers/irqchip/irq-gic-v3-its-fsl-mc-msi.c @@ -9,8 +9,6 @@ #include <linux/acpi.h> #include <linux/acpi_iort.h> -#include <linux/of_device.h> -#include <linux/of_address.h> #include <linux/irq.h> #include <linux/msi.h> #include <linux/of.h> diff --git a/drivers/irqchip/irq-imx-intmux.c b/drivers/irqchip/irq-imx-intmux.c index 80aaea82468a..6d9a08238c9d 100644 --- a/drivers/irqchip/irq-imx-intmux.c +++ b/drivers/irqchip/irq-imx-intmux.c @@ -50,8 +50,9 @@ #include <linux/irqchip/chained_irq.h> #include <linux/irqdomain.h> #include <linux/kernel.h> +#include <linux/mod_devicetable.h> #include <linux/of_irq.h> -#include <linux/of_platform.h> +#include <linux/platform_device.h> #include <linux/spinlock.h> #include <linux/pm_runtime.h> diff --git a/drivers/irqchip/irq-imx-irqsteer.c b/drivers/irqchip/irq-imx-irqsteer.c index 96230a04ec23..bd9543314539 100644 --- a/drivers/irqchip/irq-imx-irqsteer.c +++ b/drivers/irqchip/irq-imx-irqsteer.c @@ -10,8 +10,9 @@ #include <linux/irqchip/chained_irq.h> #include <linux/irqdomain.h> #include <linux/kernel.h> +#include <linux/of.h> #include <linux/of_irq.h> -#include <linux/of_platform.h> +#include <linux/platform_device.h> #include <linux/pm_runtime.h> #include <linux/spinlock.h> diff --git a/drivers/irqchip/irq-keystone.c b/drivers/irqchip/irq-keystone.c index ba9792e60329..a36396db4b08 100644 --- a/drivers/irqchip/irq-keystone.c +++ b/drivers/irqchip/irq-keystone.c @@ -15,7 +15,7 @@ #include <linux/irqdomain.h> #include <linux/irqchip.h> #include <linux/of.h> -#include <linux/of_platform.h> +#include <linux/platform_device.h> #include <linux/mfd/syscon.h> #include <linux/regmap.h> diff --git a/drivers/irqchip/irq-loongson-htvec.c b/drivers/irqchip/irq-loongson-htvec.c index fc8bf1f5d41b..0bff728b25e3 100644 --- a/drivers/irqchip/irq-loongson-htvec.c +++ b/drivers/irqchip/irq-loongson-htvec.c @@ -15,7 +15,6 @@ #include <linux/platform_device.h> #include <linux/of_address.h> #include <linux/of_irq.h> -#include <linux/of_platform.h> #include <linux/syscore_ops.h> /* Registers */ diff --git a/drivers/irqchip/irq-loongson-pch-pic.c b/drivers/irqchip/irq-loongson-pch-pic.c index 93a71f66efeb..63db8e2172e0 100644 --- a/drivers/irqchip/irq-loongson-pch-pic.c +++ b/drivers/irqchip/irq-loongson-pch-pic.c @@ -12,9 +12,9 @@ #include <linux/irqdomain.h> #include <linux/kernel.h> #include <linux/platform_device.h> +#include <linux/of.h> #include <linux/of_address.h> #include <linux/of_irq.h> -#include <linux/of_platform.h> #include <linux/syscore_ops.h> /* Registers */ diff --git a/drivers/irqchip/irq-madera.c b/drivers/irqchip/irq-madera.c index 8b81271c823c..3eb1f8cdf674 100644 --- a/drivers/irqchip/irq-madera.c +++ b/drivers/irqchip/irq-madera.c @@ -10,12 +10,10 @@ #include <linux/interrupt.h> #include <linux/irq.h> #include <linux/irqdomain.h> +#include <linux/platform_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/slab.h> -#include <linux/of.h> -#include <linux/of_device.h> -#include <linux/of_irq.h> #include <linux/irqchip/irq-madera.h> #include <linux/mfd/madera/core.h> #include <linux/mfd/madera/pdata.h> diff --git a/drivers/irqchip/irq-pruss-intc.c b/drivers/irqchip/irq-pruss-intc.c index fa8d89b02ec0..89e337ebc467 100644 --- a/drivers/irqchip/irq-pruss-intc.c +++ b/drivers/irqchip/irq-pruss-intc.c @@ -17,7 +17,7 @@ #include <linux/irqchip/chained_irq.h> #include <linux/irqdomain.h> #include <linux/module.h> -#include <linux/of_device.h> +#include <linux/of.h> #include <linux/platform_device.h> /* diff --git a/drivers/irqchip/irq-qcom-mpm.c b/drivers/irqchip/irq-qcom-mpm.c index d30614661eea..7124565234a5 100644 --- a/drivers/irqchip/irq-qcom-mpm.c +++ b/drivers/irqchip/irq-qcom-mpm.c @@ -14,7 +14,7 @@ #include <linux/mailbox_client.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_device.h> +#include <linux/of_platform.h> #include <linux/platform_device.h> #include <linux/pm_domain.h> #include <linux/slab.h> diff --git a/drivers/irqchip/irq-renesas-intc-irqpin.c b/drivers/irqchip/irq-renesas-intc-irqpin.c index 26e4c17a7bf2..fa19585f3dee 100644 --- a/drivers/irqchip/irq-renesas-intc-irqpin.c +++ b/drivers/irqchip/irq-renesas-intc-irqpin.c @@ -17,7 +17,6 @@ #include <linux/err.h> #include <linux/slab.h> #include <linux/module.h> -#include <linux/of_device.h> #include <linux/pm_runtime.h> #define INTC_IRQPIN_MAX 8 /* maximum 8 interrupts per driver instance */ diff --git a/drivers/irqchip/irq-st.c b/drivers/irqchip/irq-st.c index 819a12297b58..de71bb350d57 100644 --- a/drivers/irqchip/irq-st.c +++ b/drivers/irqchip/irq-st.c @@ -10,7 +10,7 @@ #include <dt-bindings/interrupt-controller/irq-st.h> #include <linux/err.h> #include <linux/mfd/syscon.h> -#include <linux/of_device.h> +#include <linux/of.h> #include <linux/platform_device.h> #include <linux/regmap.h> #include <linux/slab.h> diff --git a/drivers/irqchip/irq-stm32-exti.c b/drivers/irqchip/irq-stm32-exti.c index b5fa76ce5046..d8ba5fba7450 100644 --- a/drivers/irqchip/irq-stm32-exti.c +++ b/drivers/irqchip/irq-stm32-exti.c @@ -14,10 +14,11 @@ #include <linux/irqchip.h> #include <linux/irqchip/chained_irq.h> #include <linux/irqdomain.h> +#include <linux/mod_devicetable.h> #include <linux/module.h> #include <linux/of_address.h> #include <linux/of_irq.h> -#include <linux/of_platform.h> +#include <linux/platform_device.h> #include <linux/syscore_ops.h> #include <dt-bindings/interrupt-controller/arm-gic.h> diff --git a/drivers/irqchip/irq-sunxi-nmi.c b/drivers/irqchip/irq-sunxi-nmi.c index 21d49791f855..e760b1278143 100644 --- a/drivers/irqchip/irq-sunxi-nmi.c +++ b/drivers/irqchip/irq-sunxi-nmi.c @@ -19,7 +19,6 @@ #include <linux/irqdomain.h> #include <linux/of_irq.h> #include <linux/of_address.h> -#include <linux/of_platform.h> #include <linux/irqchip.h> #include <linux/irqchip/chained_irq.h> diff --git a/drivers/irqchip/irq-tb10x.c b/drivers/irqchip/irq-tb10x.c index 8a0e69298e83..680586354d12 100644 --- a/drivers/irqchip/irq-tb10x.c +++ b/drivers/irqchip/irq-tb10x.c @@ -13,7 +13,6 @@ #include <linux/irqchip.h> #include <linux/of_irq.h> #include <linux/of_address.h> -#include <linux/of_platform.h> #include <linux/io.h> #include <linux/slab.h> #include <linux/bitops.h> diff --git a/drivers/irqchip/irq-ti-sci-inta.c b/drivers/irqchip/irq-ti-sci-inta.c index 7133f9fa6fd9..b83f5cbab123 100644 --- a/drivers/irqchip/irq-ti-sci-inta.c +++ b/drivers/irqchip/irq-ti-sci-inta.c @@ -15,9 +15,9 @@ #include <linux/msi.h> #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/of_address.h> +#include <linux/of.h> #include <linux/of_irq.h> -#include <linux/of_platform.h> +#include <linux/platform_device.h> #include <linux/irqchip/chained_irq.h> #include <linux/soc/ti/ti_sci_inta_msi.h> #include <linux/soc/ti/ti_sci_protocol.h> diff --git a/drivers/irqchip/irq-ti-sci-intr.c b/drivers/irqchip/irq-ti-sci-intr.c index 1186f1e431a3..c027cd9e4a69 100644 --- a/drivers/irqchip/irq-ti-sci-intr.c +++ b/drivers/irqchip/irq-ti-sci-intr.c @@ -12,9 +12,9 @@ #include <linux/io.h> #include <linux/irqchip.h> #include <linux/irqdomain.h> -#include <linux/of_platform.h> -#include <linux/of_address.h> +#include <linux/of.h> #include <linux/of_irq.h> +#include <linux/platform_device.h> #include <linux/soc/ti/ti_sci_protocol.h> /** diff --git a/drivers/irqchip/irq-uniphier-aidet.c b/drivers/irqchip/irq-uniphier-aidet.c index 716b1bb88bf2..601f9343d5b3 100644 --- a/drivers/irqchip/irq-uniphier-aidet.c +++ b/drivers/irqchip/irq-uniphier-aidet.c @@ -12,7 +12,6 @@ #include <linux/irqdomain.h> #include <linux/kernel.h> #include <linux/of.h> -#include <linux/of_device.h> #include <linux/of_irq.h> #include <linux/platform_device.h> #include <linux/spinlock.h> diff --git a/drivers/irqchip/irqchip.c b/drivers/irqchip/irqchip.c index 7899607fbee8..1eeb0d0156ce 100644 --- a/drivers/irqchip/irqchip.c +++ b/drivers/irqchip/irqchip.c @@ -10,7 +10,7 @@ #include <linux/acpi.h> #include <linux/init.h> -#include <linux/of_device.h> +#include <linux/of.h> #include <linux/of_irq.h> #include <linux/irqchip.h> #include <linux/platform_device.h> diff --git a/drivers/irqchip/qcom-pdc.c b/drivers/irqchip/qcom-pdc.c index d96916cf6a41..a32c0d28d038 100644 --- a/drivers/irqchip/qcom-pdc.c +++ b/drivers/irqchip/qcom-pdc.c @@ -14,7 +14,6 @@ #include <linux/module.h> #include <linux/of.h> #include <linux/of_address.h> -#include <linux/of_device.h> #include <linux/of_irq.h> #include <linux/soc/qcom/irq.h> #include <linux/spinlock.h> -- 2.40.1

1 0

[PATCH v1] ASoC: tas2505: Add tas2505 driver
by Kevin Lu 14 Jul '23

14 Jul '23

Create tas2505 driver. Signed-off-by: Kevin Lu <luminlong(a)139.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2505.c | 691 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2505.h | 105 ++++++ 4 files changed, 803 insertions(+) create mode 100644 sound/soc/codecs/tas2505.c create mode 100644 sound/soc/codecs/tas2505.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8020097..2527a9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -229,6 +229,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_STA529 imply SND_SOC_STAC9766 imply SND_SOC_STI_SAS + imply SND_SOC_TAS2505 imply SND_SOC_TAS2552 imply SND_SOC_TAS2562 imply SND_SOC_TAS2764 @@ -1676,6 +1677,10 @@ config SND_SOC_STAC9766 config SND_SOC_STI_SAS tristate "codec Audio support for STI SAS codec" +config SND_SOC_TAS2505 + tristate "Texas Instruments TAS2505 Mono Audio amplifier" + depends on I2C + config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5cdbae8..95717d6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -365,6 +365,7 @@ snd-soc-max9877-objs := max9877.o snd-soc-max98504-objs := max98504.o snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-tas2505-objs := tas2505.o snd-soc-tas2552-objs := tas2552.o snd-soc-tas2562-objs := tas2562.o snd-soc-tas2764-objs := tas2764.o @@ -629,6 +630,7 @@ obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o +obj-$(CONFIG_SND_SOC_TAS2505) += snd-soc-tas2505.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o diff --git a/sound/soc/codecs/tas2505.c b/sound/soc/codecs/tas2505.c new file mode 100644 index 0000000..cf03859 --- /dev/null +++ b/sound/soc/codecs/tas2505.c @@ -0,0 +1,691 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ALSA SoC TAS2505 codec driver + * + * Author: Hieu Tran Dang <dangtranhieu2012(a)gmail.com> + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + */ +#include <linux/version.h> +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/clk.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include "tas2505.h" + +enum { + MCLK = 0, + BCLK = 1, +}; + +struct tas2505 { + int dac_vol; + int spk_vol1; +} tas2505_vol; + + +static int tas2505_dac_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = tas2505_vol.dac_vol; + + return 0; +} + +static int tas2505_dac_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + u8 val; + + val = ucontrol->value.integer.value[0]; + val = mc->invert ? mc->max - val : val; + val = (val < 0) ? 0 : val; + tas2505_vol.dac_vol = val; + snd_soc_component_write(component, TAS2505_DACVOL, val); + + return 0; +} + +static int tas2505_spkdrv_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int val; + + val = snd_soc_component_read(component, TAS2505_SPKVOL1); + + val = (val > mc->max) ? mc->max : val; + val = mc->invert ? mc->max - val : val; + ucontrol->value.integer.value[0] = val; + tas2505_vol.spk_vol1 = val; + return 0; +} + +static int tas2505_spkdrv_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + u8 val; + + val = (ucontrol->value.integer.value[0] & 0x7f); + val = mc->invert ? mc->max - val : val; + val = (val < 0) ? 0 : val; + tas2505_vol.spk_vol1 = val; + snd_soc_component_write(component, TAS2505_SPKVOL1, val); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_LINEAR(spk_drv_vol_tlv, TLV_DB_GAIN_MUTE, 0); +static const DECLARE_TLV_DB_SCALE(spk_amp_vol_tlv, 0, 600, 1); + +static const struct snd_kcontrol_new tas2505_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("DAC Playback Volume", TAS2505_DACVOL, + 0, 48, 0x81, 1, + tas2505_dac_getvol, tas2505_dac_putvol, dac_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Driver Volume", TAS2505_SPKVOL1, + 0, 0, 117, 1, + tas2505_spkdrv_getvol, tas2505_spkdrv_putvol, spk_drv_vol_tlv), + SOC_SINGLE_TLV("Speaker Amplifier Volume", TAS2505_SPKVOL2, + 4, 5, 0, spk_amp_vol_tlv), +}; + +static const struct snd_soc_dapm_widget tas2505_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC Channel", "Playback", + TAS2505_DACSETUP1, 7, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Driver", TAS2505_SPKAMPCTRL1, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("Speaker"), +}; + +static const struct snd_soc_dapm_route tas2505_audio_map[] = { + { "Speaker Driver", NULL, "DAC Channel" }, + { "Speaker", NULL, "Speaker Driver" }, +}; + +static const struct reg_default tas2505_reg_defaults[] = { + { TAS2505_CLKMUX, 0x00 }, + { TAS2505_PLLPR, 0x11 }, + { TAS2505_PLLJ, 0x04 }, + { TAS2505_PLLDMSB, 0x00 }, + { TAS2505_PLLDLSB, 0x00 }, + { TAS2505_NDAC, 0x01 }, + { TAS2505_MDAC, 0x01 }, + { TAS2505_DOSRMSB, 0x00 }, + { TAS2505_DOSRLSB, 0x80 }, + { TAS2505_IFACE1, 0x00 }, + { TAS2505_IFACE3, 0x00 }, + { TAS2505_DACSETUP1, 0x14 }, + { TAS2505_DACSETUP2, 0x0c }, + { TAS2505_DACVOL, 0x00 }, + { TAS2505_REF_POR_LDO_BGAP_CTRL, 0x00 }, + { TAS2505_LDO_CTRL, 0x0c }, + { TAS2505_SPKAMPCTRL1, 0x00 }, + { TAS2505_SPKVOL1, 0x00 }, + { TAS2505_SPKVOL2, 0x50 }, + { TAS2505_DACFLAG1, 0x00 }, + { TAS2505_DACFLAG2, 0x00 }, + { TAS2505_STICKYFLAG1, 0x00 }, + { TAS2505_STICKYFLAG2, 0x00 }, + { TAS2505_INTFLAG1, 0x00 }, + { TAS2505_INTFLAG2, 0x00 }, + { TAS2505_DACINSTRSET, 0x02 }, + { TAS2505_DACANLGAINFLAG, 0x00 }, +}; + +static bool tas2505_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2505_PAGECTL: + case TAS2505_RESET: + case TAS2505_DACFLAG1: + case TAS2505_DACFLAG2: + case TAS2505_STICKYFLAG1: + case TAS2505_STICKYFLAG2: + case TAS2505_INTFLAG1: + case TAS2505_INTFLAG2: + case TAS2505_DACANLGAINFLAG: + return true; + } + return false; +} + +static bool tas2505_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2505_DACFLAG1: + case TAS2505_DACFLAG2: + case TAS2505_STICKYFLAG1: + case TAS2505_STICKYFLAG2: + case TAS2505_INTFLAG1: + case TAS2505_INTFLAG2: + case TAS2505_DACANLGAINFLAG: + return false; + } + return true; +} + +static const struct regmap_range_cfg tas2505_ranges[] = { + { + .range_min = 0, + .range_max = 69 * 128, + .selector_reg = TAS2505_PAGECTL, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tas2505_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = tas2505_writeable, + .volatile_reg = tas2505_volatile, + .reg_defaults = tas2505_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2505_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = tas2505_ranges, + .num_ranges = ARRAY_SIZE(tas2505_ranges), + .max_register = 69 * 128, +}; + +struct tas2505_rate_divs { + u32 mclk_p; + u32 rate; + u8 pll_r; + u8 pll_j; + u16 pll_d; + u8 mdac; + u8 ndac; + u16 dosr; +}; + +static const struct tas2505_rate_divs tas2505_divs_mclksrc[] = { + { 12288000, 44100, 1, 7, 35, 4, 4, 128 }, + { 12288000, 48000, 1, 7, 0, 7, 2, 128 }, + { 2822400, 44100, 1, 8, 4672, 3, 5, 128 }, + { 11289600, 44100, 1, 8, 0, 4, 8, 64 }, + { 11289600, 22050, 1, 8, 0, 4, 8, 128 }, + { 2822400, 44100, 1, 8, 4672, 3, 5, 128 }, + { 2822400, 22050, 1, 8, 4672, 3, 5, 256 }, +}; + +static const struct tas2505_rate_divs tas2505_divs_bclksrc[] = { + { 1536000, 48000, 1, 56, 0, 2, 7, 128 }, + { 1411200, 44100, 1, 60, 0, 5, 3, 128 }, + { 3072000, 48000, 1, 28, 0, 2, 7, 128 }, + { 2822400, 44100, 1, 30, 0, 5, 3, 128 }, + { 1024000, 32000, 2, 40, 0, 5, 4, 128 }, + { 2048000, 32000, 1, 40, 0, 5, 4, 128 }, +}; + +struct tas2505_priv { + void *codec; + struct device *dev; + struct regmap *regmap; + u32 sysclk; + u32 clk_src; + u32 rate; + int rate_div_line; + int pll_clkin; + int frame_size; + u8 p_div; +}; + +static int tas2505_setup_pll(struct snd_soc_component *codec) +{ + struct tas2505_priv *tas2505 = snd_soc_component_get_drvdata(codec); + int mclk_p = tas2505->sysclk / tas2505->p_div; + struct tas2505_rate_divs *rate_divs; + int match = -1; + u8 p_div; + int i; + + if (tas2505->clk_src == MCLK) { + for (i = 0; i < ARRAY_SIZE(tas2505_divs_mclksrc); i++) { + if ( + tas2505_divs_mclksrc[i].rate == tas2505->rate && + tas2505_divs_mclksrc[i].mclk_p == mclk_p + ) { + match = i; + break; + } + } + + if (match == -1) { + dev_err(codec->dev, + "Sample rate (%u) and format not supported\n", + tas2505->rate); + return -EINVAL; + } + + tas2505->rate_div_line = match; + rate_divs = (struct tas2505_rate_divs *)tas2505_divs_mclksrc; + } else { + for (i = 0; i < ARRAY_SIZE(tas2505_divs_bclksrc); i++) { + if (tas2505_divs_bclksrc[i].rate == + tas2505->rate) { + match = i; + break; + } + } + + if (match == -1) { + dev_err(codec->dev, + "Sample rate (%u) and format not supported\n", + tas2505->rate); + return -EINVAL; + } + rate_divs = (struct tas2505_rate_divs *)tas2505_divs_bclksrc; + } + + p_div = (tas2505->p_div == 8) ? 0 : tas2505->p_div; + p_div <<= TAS2505_PLLPR_P_SHIFT; + p_div |= rate_divs[match].pll_r; + p_div |= TAS2505_PM_MASK; + + snd_soc_component_write(codec, TAS2505_PLLPR, p_div); + snd_soc_component_write(codec, TAS2505_PLLJ, + rate_divs[match].pll_j); + snd_soc_component_write(codec, TAS2505_PLLDMSB, + rate_divs[match].pll_d >> 8); + snd_soc_component_write(codec, TAS2505_PLLDLSB, + rate_divs[match].pll_d & 0xff); + mdelay(15); + snd_soc_component_write(codec, TAS2505_NDAC, + TAS2505_PM_MASK | rate_divs[match].ndac); + snd_soc_component_write(codec, TAS2505_MDAC, + TAS2505_PM_MASK | rate_divs[match].mdac); + snd_soc_component_write(codec, TAS2505_DOSRMSB, + rate_divs[match].dosr >> 8); + snd_soc_component_write(codec, TAS2505_DOSRLSB, + rate_divs[match].dosr & 0xff); + if (tas2505->clk_src != MCLK) + snd_soc_component_update_bits(codec, TAS2505_BCLKNDIV, + TAS2505_BCLKNDIV_MASK, + (rate_divs[match].dosr * rate_divs[match].mdac) / + tas2505->frame_size); + + return 0; +} + +static int tas2505_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct tas2505_priv *tas2505 = snd_soc_component_get_drvdata(codec); + u8 data = 0; + + switch (params_width(params)) { + case 16: + break; + + case 20: + data = TAS2505_WORD_LEN_20BITS; + break; + + case 24: + data = TAS2505_WORD_LEN_24BITS; + break; + + case 32: + data = TAS2505_WORD_LEN_32BITS; + break; + + default: + dev_err(codec->dev, "Unsupported width %d\n", + params_width(params)); + return -EINVAL; + } + dev_info(codec->dev, "%s: bit width = %d\n", + __func__, params_width(params)); + data <<= TAS2505_IFACE1_DATALEN_SHIFT; + + snd_soc_component_update_bits(codec, TAS2505_IFACE1, + TAS2505_IFACE1_DATALEN_MASK, data); + tas2505->rate = params_rate(params); + tas2505->frame_size = snd_soc_params_to_frame_size(params); + + return tas2505_setup_pll(codec); +} + +static int tas2505_dac_mute(struct snd_soc_dai *dai, int mute, + int stream) +{ + struct snd_soc_component *codec = dai->component; + + if (mute) { + snd_soc_component_write(codec, TAS2505_DACVOL, 0x81); + snd_soc_component_update_bits(codec, TAS2505_DACSETUP2, + TAS2505_DACSETUP2_MUTE_MASK, + TAS2505_DACSETUP2_MUTE_MASK); + mdelay(25); + snd_soc_component_update_bits(codec, TAS2505_DACSETUP1, 0x80, 0x00); + } else { + snd_soc_component_update_bits(codec, TAS2505_DACSETUP1, 0x80, 0x80); + mdelay(25); + snd_soc_component_write(codec, TAS2505_DACVOL, tas2505_vol.dac_vol); + snd_soc_component_update_bits(codec, TAS2505_DACSETUP2, + TAS2505_DACSETUP2_MUTE_MASK, 0x0); + } + + return 0; +} + +static int tas2505_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *codec = codec_dai->component; + u8 iface_reg1 = 0; + u8 iface_reg3 = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg1 |= TAS2505_IFACE1_BCLKDIR_MASK; + iface_reg1 |= TAS2505_IFACE1_WCLKDIR_MASK; + break; + + default: + dev_err(codec->dev, + "%s SND_SOC_DAIFMT_MASTER_MASK 0x%x\n", + __func__, fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + + case SND_SOC_DAIFMT_IB_NF: + iface_reg3 |= TAS2505_IFACE3_BCLKINV_MASK; + break; + + default: + dev_err(codec->dev, + "%s SND_SOC_DAIFMT_INV_MASK 0x%x\n", + __func__, fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + iface_reg1 |= (TAS2505_DSP_MODE << + TAS2505_IFACE1_INTERFACE_SHIFT); + break; + + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg1 |= (TAS2505_RJF_MODE << + TAS2505_IFACE1_INTERFACE_SHIFT); + break; + + case SND_SOC_DAIFMT_LEFT_J: + iface_reg1 |= (TAS2505_LJF_MODE << + TAS2505_IFACE1_INTERFACE_SHIFT); + break; + + default: + dev_err(codec->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_component_write(codec, TAS2505_IFACE1, iface_reg1); + snd_soc_component_update_bits(codec, TAS2505_IFACE3, + TAS2505_IFACE3_BCLKINV_MASK | TAS2505_IFACE3_BDIVCLKIN_MASK, + iface_reg3); + + return 0; +} + +static int tas2505_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct tas2505_priv *tas2505 = snd_soc_component_get_drvdata(codec); + int i, x; + int ret = 0; + + dev_info(tas2505->dev, "%s: clk_id: %d, freq: %d\n", + __func__, clk_id, freq); + + if (tas2505->clk_src == MCLK) { + for (i = 0; i < ARRAY_SIZE(tas2505_divs_mclksrc); i++) { + for (x = 1; x < 9; x++) { + if ((freq / x) == tas2505_divs_mclksrc[i].mclk_p) { + tas2505->p_div = x; + break; + } + } + if (x < 9) + break; + } + + if (i != ARRAY_SIZE(tas2505_divs_mclksrc)) { + } else { + dev_err(tas2505->dev, + "Can't produce required PLL_CLKIN frequency\n"); + ret = -EINVAL; + } + tas2505->pll_clkin = clk_id; + } else { + tas2505->pll_clkin = 1; + tas2505->p_div = 1; + } + + if (!ret) { + snd_soc_component_write(codec, TAS2505_CLKMUX, + (tas2505->pll_clkin << TAS2505_PLL_CLKIN_SHIFT) | + TAS2505_CODEC_CLKIN_PLL); + + tas2505->sysclk = freq; + } + return ret; + +} + +static void tas2505_power_on(struct snd_soc_component *codec) +{ + snd_soc_component_update_bits(codec, TAS2505_LDO_CTRL, + TAS2505_LDO_PLL_HP_LVL_MASK, 0); + snd_soc_component_update_bits(codec, TAS2505_REF_POR_LDO_BGAP_CTRL, + TAS2505_REF_POR_LDO_BGAP_MASTER_REF_MASK, + TAS2505_REF_POR_LDO_BGAP_MASTER_REF_MASK); +} + +static void tas2505_power_off(struct snd_soc_component *codec) +{ + snd_soc_component_update_bits(codec, TAS2505_REF_POR_LDO_BGAP_CTRL, + TAS2505_REF_POR_LDO_BGAP_MASTER_REF_MASK, 0); + snd_soc_component_update_bits(codec, TAS2505_LDO_CTRL, + TAS2505_LDO_PLL_HP_LVL_MASK, + TAS2505_LDO_PLL_HP_LVL_MASK); +} + +static int tas2505_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + int current_lvl = snd_soc_component_get_bias_level(codec); + int rc = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (current_lvl == SND_SOC_BIAS_OFF) { + tas2505_power_on(codec); + dev_info(codec->dev, + "%s:power on\n", __func__); + } else + dev_info(codec->dev, + "%s:SND_SOC_BIAS_STANDBY\n", + __func__); + break; + case SND_SOC_BIAS_OFF: + if (current_lvl == SND_SOC_BIAS_STANDBY) { + tas2505_power_off(codec); + dev_info(codec->dev, + "%s:power off\n", + __func__); + } else + dev_info(codec->dev, + "%s:SND_SOC_BIAS_OFF\n", + __func__); + break; + default: + dev_err(codec->dev, "Invalid bias level\n"); + rc = -EINVAL; + } + + return rc; +} + +static int tas2505_codec_probe(struct snd_soc_component *codec) +{ + struct tas2505_priv *tas2505 = snd_soc_component_get_drvdata(codec); + + tas2505->codec = codec; + tas2505_vol.dac_vol = 0x00; + tas2505_vol.spk_vol1 = 0x00; + + snd_soc_component_write(codec, TAS2505_RESET, 1); + + return 0; +} + +static const struct snd_soc_component_driver soc_codec_driver_tas2505 = { + .probe = tas2505_codec_probe, + .set_bias_level = tas2505_set_bias_level, + .suspend_bias_off = true, + + .controls = tas2505_snd_controls, + .num_controls = ARRAY_SIZE(tas2505_snd_controls), + .dapm_widgets = tas2505_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2505_dapm_widgets), + .dapm_routes = tas2505_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2505_audio_map), +}; + +static const struct snd_soc_dai_ops tas2505_dai_ops = { + .hw_params = tas2505_hw_params, + .set_sysclk = tas2505_set_dai_sysclk, + .set_fmt = tas2505_set_dai_fmt, + .mute_stream = tas2505_dac_mute, +}; + +static struct snd_soc_dai_driver tas2505_dai_driver[] = { + { + .name = "tas2505-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = TAS2505_RATES, + .formats = TAS2505_FORMATS, + }, + .ops = &tas2505_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tas2505_i2c_probe(struct i2c_client *i2c) +{ + struct tas2505_priv *tas2505; + struct device_node *np = i2c->dev.of_node; + const struct regmap_config *regmap_config = &tas2505_i2c_regmap; + int ret; + + tas2505 = devm_kzalloc(&i2c->dev, sizeof(*tas2505), GFP_KERNEL); + if (tas2505 == NULL) + return -ENOMEM; + + tas2505->regmap = devm_regmap_init_i2c(i2c, regmap_config); + if (IS_ERR(tas2505->regmap)) { + ret = PTR_ERR(tas2505->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + ret = of_get_named_gpio(np, "gpio-reset", 0); + if ((ret > 0) && gpio_is_valid(ret)) { + devm_gpio_request_one(&i2c->dev, ret, GPIOF_OUT_INIT_HIGH, + "reset"); + } + + ret = of_property_read_u32(np, "clock-source", + &tas2505->clk_src); + + dev_info(&i2c->dev, "%s: clock-source = %s\n", __func__, + tas2505->clk_src?"BCLK":"MCLK"); + + tas2505->dev = &i2c->dev; + + dev_set_drvdata(tas2505->dev, tas2505); + + return snd_soc_register_component(&i2c->dev, + &soc_codec_driver_tas2505, tas2505_dai_driver, + ARRAY_SIZE(tas2505_dai_driver)); +} + +static void tas2505_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); +} + +static const struct of_device_id tas2505_of_match[] = { + { .compatible = "ti,tas2505" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2505_of_match); + +static const struct i2c_device_id tas2505_i2c_id[] = { + { "tas2505", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2ic, tas2505_i2c_id); + +static struct i2c_driver tas2505_i2c_driver = { + .driver = { + .name = "tas2505-codec", + .of_match_table = of_match_ptr(tas2505_of_match), + }, + .probe = tas2505_i2c_probe, + .remove = tas2505_i2c_remove, + .id_table = tas2505_i2c_id, +}; + +module_i2c_driver(tas2505_i2c_driver); + +MODULE_DESCRIPTION("ASoC TAS2505 codec driver"); +MODULE_AUTHOR("Hieu Tran Dang <dangtranhieu2012(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2505.h b/sound/soc/codecs/tas2505.h new file mode 100644 index 0000000..79c3f84 --- /dev/null +++ b/sound/soc/codecs/tas2505.h @@ -0,0 +1,105 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * ALSA SoC TAS2505 codec driver + * + * Author: Hieu Tran Dang <dangtranhieu2012(a)gmail.com> + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + */ + +#ifndef _TAS2505_H +#define _TAS2505_H + +#define TAS2505_RATES (SNDRV_PCM_RATE_8000_96000) +#define TAS2505_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define TAS2505_REG(page, reg) ((page * 128) + reg) + +#define TAS2505_PAGECTL TAS2505_REG(0, 0) +#define TAS2505_RESET TAS2505_REG(0, 1) +#define TAS2505_CLKMUX TAS2505_REG(0, 4) +#define TAS2505_PLLPR TAS2505_REG(0, 5) +#define TAS2505_PLLJ TAS2505_REG(0, 6) +#define TAS2505_PLLDMSB TAS2505_REG(0, 7) +#define TAS2505_PLLDLSB TAS2505_REG(0, 8) +#define TAS2505_NDAC TAS2505_REG(0, 11) +#define TAS2505_MDAC TAS2505_REG(0, 12) +#define TAS2505_DOSRMSB TAS2505_REG(0, 13) +#define TAS2505_DOSRLSB TAS2505_REG(0, 14) +#define TAS2505_IFACE1 TAS2505_REG(0, 27) +#define TAS2505_IFACE3 TAS2505_REG(0, 29) +#define TAS2505_BCLKNDIV TAS2505_REG(0, 30) +#define TAS2505_DACFLAG1 TAS2505_REG(0, 37) +#define TAS2505_DACFLAG2 TAS2505_REG(0, 38) +#define TAS2505_STICKYFLAG1 TAS2505_REG(0, 42) +#define TAS2505_INTFLAG1 TAS2505_REG(0, 43) +#define TAS2505_STICKYFLAG2 TAS2505_REG(0, 44) +#define TAS2505_INTFLAG2 TAS2505_REG(0, 46) +#define TAS2505_DACINSTRSET TAS2505_REG(0, 60) +#define TAS2505_DACSETUP1 TAS2505_REG(0, 63) +#define TAS2505_DACSETUP2 TAS2505_REG(0, 64) +#define TAS2505_DACVOL TAS2505_REG(0, 65) +#define TAS2505_REF_POR_LDO_BGAP_CTRL TAS2505_REG(1, 1) +#define TAS2505_LDO_CTRL TAS2505_REG(1, 2) +#define TAS2505_PLAYBACKCONF1 TAS2505_REG(1, 3) +#define TAS2505_SPKAMPCTRL1 TAS2505_REG(1, 45) +#define TAS2505_SPKVOL1 TAS2505_REG(1, 46) +#define TAS2505_SPKVOL2 TAS2505_REG(1, 48) +#define TAS2505_DACANLGAINFLAG TAS2505_REG(1, 63) + +#define TAS2505_PLLPR_P_MASK (0x70) +#define TAS2505_PLLPR_R_MASK (0xf) +#define TAS2505_PLL_DAC_MASK (0x7f) +#define TAS2505_BCLKNDIV_MASK (0x7f) +#define TAS2505_IFACE1_DATALEN_MASK (0x30) +#define TAS2505_IFACE1_WCLKDIR_MASK (0x4) +#define TAS2505_IFACE1_BCLKDIR_MASK (0x8) +#define TAS2505_IFACE1_INTERFACE_MASK (0xc0) +#define TAS2505_IFACE3_BDIVCLKIN_MASK (0x1) +#define TAS2505_IFACE3_BCLKINV_MASK (0x8) +#define TAS2505_DACSETUP1_PATH_CTRL_MSK (0x30) +#define TAS2505_DACSETUP2_MUTE_MASK (0x8) +#define TAS2505_PM_MASK (0x80) +#define TAS2505_LDO_PLL_HP_LVL_MASK (0x8) +#define TAS2505_REF_POR_LDO_BGAP_MASTER_REF_MASK (0x10) +#define TAS2505_SPKVOL2_MSK (0x70) +#define TAS2505_CODEC_CLKIN_MSK (0x3) +#define TAS2505_PLL_INPUT_CLK_MSK (0xC) +#define TAS2505_SPKAMPCTRL1_SPKDRV_MSK BIT(1) + +#define TAS2505_PLLPR_P_SHIFT (4) +#define TAS2505_PLL_CLKIN_SHIFT (2) +#define TAS2505_IFACE1_DATALEN_SHIFT (4) +#define TAS2505_IFACE1_INTERFACE_SHIFT (6) +#define TAS2505_IFACE3_BCLKINV_SHIFT (4) +#define TAS2505_SPKVOL2_6DB_SHIFT (4) +#define TAS2505_DACSETUP1_PATH_CTRL_SHIFT (4) + +#define TAS2505_WORD_LEN_20BITS (1) +#define TAS2505_WORD_LEN_24BITS (2) +#define TAS2505_WORD_LEN_32BITS (3) + +#define TAS2505_DSP_MODE (1) +#define TAS2505_RJF_MODE (2) +#define TAS2505_LJF_MODE (3) + +#define TAS2505_PLL_CLKIN_MCLK (0) +#define TAS2505_PLL_CLKIN_BCLK (1) +#define TAS2505_PLL_CLKIN_GPIO (2) +#define TAS2505_PLL_CLKIN_DIN (3) +#define TAS2505_CODEC_CLKIN_PLL (3) +#define TAS2505_SPKVOL2_MUTE (0) +#define TAS2505_SPKVOL2_6DB (1) +#define TAS2505_DACSETUP1_PATH_CTRL_LRDIV2 (3) +#define TAS2505_SPKAMPCTRL1_SPKDRV_PWD (0) +#define TAS2505_SPKAMPCTRL1_SPKDRV_PWU (1) + +#endif -- 2.34.1

3 2

[PATCH v3 00/15] PCI: Define Intel PCI IDs and use them in drivers
by Amadeusz Sławiński 14 Jul '23

14 Jul '23

PCI IDs for Intel HDA are duplicated across quite a few drivers, due to various configurations and historical reasons. Currently almost all uses of HDA PCI IDs have corresponding comment telling which platform it is. Additionally there are some inconsistencies between drivers about which ID corresponds to which device. Simplify things, by adding PCI IDs to global header and make use of them in drivers. This allows for removal of comments by having IDs themselves being self explanatory. Additionally it allows for removal of existing inconsistencies by having one source of truth. Changes from v2: - Change CHV to BSW (Andy) - Fix incorrectly rebased patches (Andy) - Fix commit message and add suggestions from Andy to SST patch (Andy) - Rebased on top of Linus tree commit: 4b810bf037e5 Changes from v1: - Remove BXT-T PCI ID as it is not released (Andy) - Reorder macros to start from pci_match_id() (Andy) - Add comment about BXT->APL name change in commit messages (Andy) - Use SST as part of macro name for DSP only devices (Andy) - Add PCI IDs for all SST combinations (Andy) Changes from RFC: - Sort Intel PCI IDs before adding new ones - Fix ordering of new PCI IDs (Andy) - Define all used Intel IDs (Andy) - Add macros for controller type detection (Andy/Bjorn) - Add set of patches changing to use above macro (Andy/Bjorn) - Use PCI_DEVICE_DATA for Intel IDs in sound/pci/hda/hda_intel.c (Andy) - Commit message wording (Andy) - Remove unnecessary tabs (Andy) Amadeusz Sławiński (15): PCI: Sort Intel PCI IDs by number PCI: Add Intel Audio DSP devices to pci_ids.h ASoC: SOF: Remove unused Broxton PCI ID ALSA: Remove unused Broxton PCI ID ALSA: hda: Add controller matching macros ALSA: hda: Use global PCI match macro ALSA: hda/i915: Use global PCI match macro ASoC: Intel: Skylake: Use global PCI match macro ALSA: intel-dsp-config: Convert to PCI device IDs defines ALSA: hda: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: Skylake: Convert to PCI device IDs defines ASoC: SOF: Intel: Convert to PCI device IDs defines ASoC: Intel: sst: Convert to PCI device IDs defines include/linux/pci_ids.h | 105 +++++-- include/sound/hda_codec.h | 3 - include/sound/hdaudio.h | 26 ++ sound/hda/hdac_i915.c | 7 +- sound/hda/intel-dsp-config.c | 124 ++++---- sound/pci/hda/hda_intel.c | 373 ++++++++++--------------- sound/soc/intel/atom/sst/sst.c | 9 +- sound/soc/intel/atom/sst/sst.h | 7 +- sound/soc/intel/atom/sst/sst_pci.c | 4 +- sound/soc/intel/avs/board_selection.c | 10 +- sound/soc/intel/avs/core.c | 16 +- sound/soc/intel/skylake/skl-messages.c | 16 +- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/intel/skylake/skl.c | 36 +-- sound/soc/sof/intel/pci-apl.c | 8 +- sound/soc/sof/intel/pci-cnl.c | 15 +- sound/soc/sof/intel/pci-icl.c | 12 +- sound/soc/sof/intel/pci-mtl.c | 3 +- sound/soc/sof/intel/pci-skl.c | 6 +- sound/soc/sof/intel/pci-tgl.c | 45 +-- sound/soc/sof/intel/pci-tng.c | 3 +- 21 files changed, 386 insertions(+), 445 deletions(-) -- 2.34.1

5 23

[PATCH v2 00/15] PCI: Define Intel PCI IDs and use them in drivers
by Amadeusz Sławiński 14 Jul '23

14 Jul '23

PCI IDs for Intel HDA are duplicated across quite a few drivers, due to various configurations and historical reasons. Currently almost all uses of HDA PCI IDs have corresponding comment telling which platform it is. Additionally there are some inconsistencies between drivers about which ID corresponds to which device. Simplify things, by adding PCI IDs to global header and make use of them in drivers. This allows for removal of comments by having IDs themselves being self explanatory. Additionally it allows for removal of existing inconsistencies by having one source of truth. Changes from v1: - Remove BXT-T PCI ID as it is not released (Andy) - Reorder macros to start from pci_match_id() (Andy) - Add comment about BXT->APL name change in commit messages (Andy) - Use SST as part of macro name for DSP only devices (Andy) - Add PCI IDs for all SST combinations (Andy) Changes from RFC: - Sort Intel PCI IDs before adding new ones - Fix ordering of new PCI IDs (Andy) - Define all used Intel IDs (Andy) - Add macros for controller type detection (Andy/Bjorn) - Add set of patches changing to use above macro (Andy/Bjorn) - Use PCI_DEVICE_DATA for Intel IDs in sound/pci/hda/hda_intel.c (Andy) - Commit message wording (Andy) - Remove unnecessary tabs (Andy) Amadeusz Sławiński (15): PCI: Sort Intel PCI IDs by number PCI: Add Intel Audio DSP devices to pci_ids.h ASoC: SOF: Remove unused Broxton PCI ID ALSA: Remove unused Broxton PCI ID ALSA: hda: Add controller matching macros ALSA: hda: Use global PCI match macro ALSA: hda/i915: Use global PCI match macro ASoC: Intel: Skylake: Use global PCI match macro ALSA: intel-dsp-config: Convert to PCI device IDs defines ALSA: hda: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: Skylake: Convert to PCI device IDs defines ASoC: SOF: Intel: Convert to PCI device IDs defines ASoC: Intel: sst: Convert to PCI device IDs defines include/linux/pci_ids.h | 105 +++++-- include/sound/hda_codec.h | 3 - include/sound/hdaudio.h | 26 ++ sound/hda/hdac_i915.c | 7 +- sound/hda/intel-dsp-config.c | 124 ++++---- sound/pci/hda/hda_intel.c | 373 ++++++++++--------------- sound/soc/intel/atom/sst/sst.c | 7 +- sound/soc/intel/atom/sst/sst.h | 3 - sound/soc/intel/atom/sst/sst_pci.c | 4 +- sound/soc/intel/avs/board_selection.c | 10 +- sound/soc/intel/avs/core.c | 16 +- sound/soc/intel/skylake/skl-messages.c | 16 +- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/intel/skylake/skl.c | 36 +-- sound/soc/sof/intel/pci-apl.c | 8 +- sound/soc/sof/intel/pci-cnl.c | 15 +- sound/soc/sof/intel/pci-icl.c | 12 +- sound/soc/sof/intel/pci-mtl.c | 3 +- sound/soc/sof/intel/pci-skl.c | 6 +- sound/soc/sof/intel/pci-tgl.c | 45 +-- sound/soc/sof/intel/pci-tng.c | 3 +- 21 files changed, 383 insertions(+), 442 deletions(-) -- 2.34.1

3 23

[PATCH] lib: Introduce KUnit tests for minmax
by Wojciech Ziemba 14 Jul '23

14 Jul '23

This patch adds KUnit coverage for minmax_array() macros from <linux/minmax.h>. It assures that these macros compile correctly and work as expected when different types are used. The test cases exercise types: s8, u8, s16, u16, s32, u32, s64, u64 with arrays of random numbers. Signed-off-by: Wojciech Ziemba <wojciech.ziemba(a)intel.com> --- lib/Kconfig.debug | 12 +++++ lib/Makefile | 1 + lib/minmax_kunit.c | 132 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 145 insertions(+) create mode 100644 lib/minmax_kunit.c diff --git a/lib/Kconfig.debug b/lib/Kconfig.debug index fbc89baf7de6..e845876482fb 100644 --- a/lib/Kconfig.debug +++ b/lib/Kconfig.debug @@ -2777,6 +2777,18 @@ config SIPHASH_KUNIT_TEST This is intended to help people writing architecture-specific optimized versions. If unsure, say N. +config MINMAX_KUNIT_TEST + tristate "KUnit test for minmax" if !KUNIT_ALL_TESTS + depends on KUNIT + default KUNIT_ALL_TESTS + help + Builds KUnit tests module for minmax. + + For more information on KUnit and unit tests in general please refer + to the KUnit documentation in Documentation/dev-tools/kunit/. + + If unsure, say N. + config TEST_UDELAY tristate "udelay test driver" help diff --git a/lib/Makefile b/lib/Makefile index 42d307ade225..950a3f388e6f 100644 --- a/lib/Makefile +++ b/lib/Makefile @@ -396,6 +396,7 @@ obj-$(CONFIG_FORTIFY_KUNIT_TEST) += fortify_kunit.o obj-$(CONFIG_STRCAT_KUNIT_TEST) += strcat_kunit.o obj-$(CONFIG_STRSCPY_KUNIT_TEST) += strscpy_kunit.o obj-$(CONFIG_SIPHASH_KUNIT_TEST) += siphash_kunit.o +obj-$(CONFIG_MINMAX_KUNIT_TEST) += minmax_kunit.o obj-$(CONFIG_GENERIC_LIB_DEVMEM_IS_ALLOWED) += devmem_is_allowed.o diff --git a/lib/minmax_kunit.c b/lib/minmax_kunit.c new file mode 100644 index 000000000000..00ec175156d6 --- /dev/null +++ b/lib/minmax_kunit.c @@ -0,0 +1,132 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Copyright (c) 2023 Intel Corporation + * + * KUnit test cases for minmax. + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <kunit/test.h> + +#include <linux/kernel.h> +#include <linux/minmax.h> +#include <linux/module.h> +#include <linux/types.h> + +#define MINMAX_ARR_TEST(type, op, expected) \ + static const type expected_##op##_##type = expected; \ +static void op##_array_##type##_test(struct kunit *test) \ +{ \ + type value = op##_array(test_vector_##type, \ + ARRAY_SIZE(test_vector_##type)); \ + KUNIT_EXPECT_EQ(test, value, expected_##op##_##type); \ +} + +/* type 's8' */ +static const s8 test_vector_s8[] = { + 1, 85, -112, -61, 6, 0, 117, -55, 68, -55, 23, 1, -100, 55, -118 +}; + +MINMAX_ARR_TEST(s8, min, -118) +MINMAX_ARR_TEST(s8, max, 117) + +/* type 'u8' */ +static const u8 test_vector_u8[] = { + 209, 145, 131, 180, 88, 97, 240, 170, 148, 158, 35, 226, 14, 244, 2 +}; + +MINMAX_ARR_TEST(u8, min, 2) +MINMAX_ARR_TEST(u8, max, 244) + +/* type 's16' */ +static const s16 test_vector_s16[] = { + 2641, 30296, -15090, 11371, 13995, 28244, 27482, -7239, -9036, -3147, + 22065, 2097, -26854, -8633, 13366 +}; + +MINMAX_ARR_TEST(s16, min, -26854) +MINMAX_ARR_TEST(s16, max, 30296) + +/* type 'u16' */ +static const u16 test_vector_u16[] = { + 50102, 58387, 42350, 33531, 32109, 65525, 25985, 31889, 14772, 7296, + 48242, 53340, 33594, 19856, 45544 +}; + +MINMAX_ARR_TEST(u16, min, 7296) +MINMAX_ARR_TEST(u16, max, 65525) + +/* type 's32' */ +static const s32 test_vector_s32[] = { + 979885732L, 885950778L, -1450966230L, -1624192399L, -1753023734L, + 530868381L, 656215973L, 630677997L, 391000329L, 1253640357L, + 438040531L, -993355872L, 585127620L, -1068766240L, 147051568L +}; + +MINMAX_ARR_TEST(s32, min, -1753023734L) +MINMAX_ARR_TEST(s32, max, 1253640357L) + +/* type 'u32' */ +static const u32 test_vector_u32[] = { + 2161504111UL, 1183092838UL, 1728689175UL, 2309150449UL, 3735269177UL, + 1550306875UL, 2147576091UL, 4187089172UL, 545942284UL, 1969608375UL, + 2624877400UL, 3029665375UL, 1293120185UL, 3801307680UL, 1161238802UL +}; + +MINMAX_ARR_TEST(u32, min, 545942284UL) +MINMAX_ARR_TEST(u32, max, 4187089172UL) + +/* type 's64' */ +static const s64 test_vector_s64[] = { + -8438490804591620000LL, 8025377321703520000LL, 3208690154945970000LL, + -2321767394263770000LL, -1278489889635390000LL, 2190470827346570000LL, + -2022822382784580000LL, 6599789147713410000LL, 3949996236706410000LL, + -8261935734503760000LL, 661645293834217000LL, 8346383784731440000LL, + -3443819189603500000LL, 7985744187902840000LL, 5083298007916820000LL +}; + +MINMAX_ARR_TEST(s64, min, -8438490804591620000LL) +MINMAX_ARR_TEST(s64, max, 8346383784731440000LL) + +/* type 'u64' */ +static const u64 test_vector_u64[] = { + 8249683091780880000ULL, 13192984997103100000ULL, 17673970905507700000ULL, + 11101419446779000000ULL, 7059417167478820000ULL, 17488355550255500000ULL, + 1592214570100350000ULL, 4324504718429290000ULL, 9233486922226120000ULL, + 17205873265072200000ULL, 15951973546886800000ULL, 8694358874973410000ULL, + 11857168979503600000ULL, 15015693759541800000ULL, 4758196429468010000ULL +}; + +MINMAX_ARR_TEST(u64, min, 1592214570100350000ULL) +MINMAX_ARR_TEST(u64, max, 17673970905507700000ULL) +#undef MINMAX_ARR_TEST + +static struct kunit_case minmax_array_test_cases[] = { + KUNIT_CASE(min_array_s8_test), + KUNIT_CASE(max_array_s8_test), + KUNIT_CASE(min_array_u8_test), + KUNIT_CASE(max_array_u8_test), + KUNIT_CASE(min_array_s16_test), + KUNIT_CASE(max_array_s16_test), + KUNIT_CASE(min_array_u16_test), + KUNIT_CASE(max_array_u16_test), + KUNIT_CASE(min_array_s32_test), + KUNIT_CASE(max_array_s32_test), + KUNIT_CASE(min_array_u32_test), + KUNIT_CASE(max_array_u32_test), + KUNIT_CASE(min_array_s64_test), + KUNIT_CASE(max_array_s64_test), + KUNIT_CASE(min_array_u64_test), + KUNIT_CASE(max_array_u64_test), + {} +}; + +static struct kunit_suite minmax_test_suite = { + .name = "minmax_array", + .test_cases = minmax_array_test_cases, +}; +kunit_test_suite(minmax_test_suite); + +MODULE_AUTHOR("Intel"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("KUnit test module for minmax"); -- 2.38.1

1 0

[PATCH v2 1/2] ALSA: pcmtest: Add 'open' PCM callback error injection
by Ivan Orlov 14 Jul '23

14 Jul '23

Extend 'pcmtest' virtual driver with 'open' callback error injection functionality, as it already can inject errors into other PCM callbacks. Add module parameter which enables EBUSY error injection in the 'open' PCM callback. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Split changes in the patch into two different patches sound/drivers/pcmtest.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 291e7fe47893..e74c523e49eb 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -65,6 +65,7 @@ static int inject_delay; static bool inject_hwpars_err; static bool inject_prepare_err; static bool inject_trigger_err; +static bool inject_open_err; static short fill_mode = FILL_MODE_PAT; @@ -88,6 +89,9 @@ module_param(inject_prepare_err, bool, 0600); MODULE_PARM_DESC(inject_prepare_err, "Inject EINVAL error in the 'prepare' callback"); module_param(inject_trigger_err, bool, 0600); MODULE_PARM_DESC(inject_trigger_err, "Inject EINVAL error in the 'trigger' callback"); +module_param(inject_open_err, bool, 0600); +MODULE_PARM_DESC(inject_open_err, "Inject EBUSY error in the 'open' callback"); + struct pcmtst { struct snd_pcm *pcm; @@ -364,6 +368,9 @@ static int snd_pcmtst_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct pcmtst_buf_iter *v_iter; + if (inject_open_err) + return -EBUSY; + v_iter = kzalloc(sizeof(*v_iter), GFP_KERNEL); if (!v_iter) return -ENOMEM; -- 2.34.1

2 2

Re: [PATCH 2/4] ASoC: codecs: es8326: Fix power-up sequence
by Zhu Ning 14 Jul '23

14 Jul '23

> > > This looks wrong, you're resyncing the cache and then start resetting > > > registers? It feels like the ordering is off here, and some of this > > > reset sequence might want to be done with the cache bypassed. Are you > > > sure that there's no corruption of user visible state resulting from the > > > power up sequence, especially around the HP driver? > > > Basically the chip runs through the start-up sequence every time in > > the resume function. > > Will move the reset sequence to es8326_suspend > > That's not going to overwrite any user visible settings? Registers written here are internal bias/vmid which should be fixed. User settings like alc and volume won't be overwritten here.

1 0

[PATCH] ASoC: simple-card: add trigger-stop entry parser
by Maxim Kochetkov 14 Jul '23

14 Jul '23

It may be useful to specify trigger-stop for some DMA-based simple audio card. So add this "trigger-stop" device tree entry parser. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/generic/simple-card.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index a78babf44f38..86df7d326068 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -145,8 +145,27 @@ static int simple_parse_node(struct asoc_simple_priv *priv, struct snd_soc_dai_link_component *dlc; struct asoc_simple_dai *dai; int ret; + const char *str; + struct { + char *name; + unsigned int val; + } of_trigger_table[] = { + { "default", SND_SOC_TRIGGER_ORDER_DEFAULT }, + { "ldc", SND_SOC_TRIGGER_ORDER_LDC }, + }; + if (cpu) { + ret = of_property_read_string(np, "trigger-stop", &str); + if (ret == 0) { + for (int i = 0; i < ARRAY_SIZE(of_trigger_table); i++) { + if (strcmp(str, of_trigger_table[i].name) == 0) { + dai_link->trigger_stop = of_trigger_table[i].val; + break; + } + } + } + dlc = asoc_link_to_cpu(dai_link, 0); dai = simple_props_to_dai_cpu(dai_props, 0); } else { -- 2.40.1

2 1

[PATCH 0/2] kselftest/alsa: Decrease pcm-test duration to avoid timeouts
by Nícolas F. R. A. Prado 13 Jul '23

13 Jul '23

This series decreases the pcm-test duration in order to avoid timeouts by first moving the audio stream duration to a variable and subsequently decreasing it. Nícolas F. R. A. Prado (2): kselftest/alsa: pcm-test: Move stream duration and margin to variables kselftest/alsa: pcm-test: Decrease stream duration from 4 to 2 seconds tools/testing/selftests/alsa/pcm-test.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) -- 2.41.0

5 15

[PATCH] ASoC: q6dsp: q6apm: make g_apm static
by Min-Hua Chen 13 Jul '23

13 Jul '23

This patch fixes the following sprse warning: sound/soc/qcom/qdsp6/q6apm.c:30:14: sparse: warning: symbol 'g_apm' was not declared. Should it be static? No functional change intended Signed-off-by: Min-Hua Chen <minhuadotchen(a)gmail.com> --- sound/soc/qcom/qdsp6/q6apm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 7bfac9492ab5..d5ad9cbd3c21 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -27,7 +27,7 @@ struct apm_graph_mgmt_cmd { #define APM_GRAPH_MGMT_PSIZE(p, n) ALIGN(struct_size(p, sub_graph_id_list, n), 8) -struct q6apm *g_apm; +static struct q6apm *g_apm; int q6apm_send_cmd_sync(struct q6apm *apm, struct gpr_pkt *pkt, uint32_t rsp_opcode) { -- 2.34.1

2 1

[PATCH 00/10] ASoC: ad: Update Analog Devices drivers to maple tree register cache
by Mark Brown 13 Jul '23

13 Jul '23

The maple tree register cache has now got to feature parity with the rbtree cache, there are some different tradeoffs made and it should be a better choice for most modern systems. Convert the Analog Devices drivers to use the more modern data structure. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (10): ASoC: ad1836: Update to use maple tree register cache ASoC: ad1980: Update to use maple tree register cache ASoC: adau1372: Update to use maple tree register cache ASoC: adau1373: Update to use maple tree register cache ASoC: adau1701: Update to use maple tree register cache ASoC: adau1761: Update to use maple tree register cache ASoC: adau1781: Update to use maple tree register cache ASoC: adau1977: Update to use maple tree register cache ASoC: adau7118: Update to use maple tree register cache ASoC: adav80x: Update to use maple tree register cache sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/adau1372.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adau1761.c | 2 +- sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau1977.c | 2 +- sound/soc/codecs/adau7118-i2c.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230701-asoc-ad-maple-170068cf0c39 Best regards, -- Mark Brown <broonie(a)kernel.org>

2 12

[PATCH 2/2] ALSA: hda/realtek: Add quirks for Lenovo 16IAH7, 16IAX7 and 16ARHA7
by David Xu 13 Jul '23

13 Jul '23

Add quirks for Lenovo 16IAH7, 16IAX7 and 16ARHA7 that have dual cs35l41 amplifiers that drives their speakers. Please note that this patch depends on patch 1, for the added quirks to work for the specified models, proper CSC3551 configurtion fixups have to be applied. Signed-off-by: David Xu <xuwd1(a)hotmail.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e2f8b608de82..cc10bb8b75d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9831,6 +9831,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3803, "Legion Slim 7i 16IAH7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), @@ -9846,6 +9847,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x386e, "Legion Slim 7i 16IAH7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x386f, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3874, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3877, "Legion 7 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.41.0

1 0

Re: [PATCH 2/4] ASoC: codecs: es8326: Fix power-up sequence
by Zhu Ning 13 Jul '23

13 Jul '23

> > Old power-up sequence causes large pop noise during start-up. > > Using a new sequence instead. > > > static const struct regmap_range es8326_volatile_ranges[] = { > > - regmap_reg_range(ES8326_HP_DETECT, ES8326_HP_DETECT), > > + regmap_reg_range(ES8326_HPDET_TYPE, ES8326_HPDET_TYPE), > > }; > > As well as not seeming to correspond to the patch description this seems > obviously buggy - even if _HPDET_TYPE was missed from the list of > volatile registers no software change should be able to make a volatile > register non-volatile, that's a property of the hardware. The changes > to read from HPDET_TYPE also seem unrelated to the main change here in > the resume function, this should probably be multiple commits. Just a naming space change, will create a new patch for it. > > static int es8326_resume(struct snd_soc_component *component) > > { > > struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); > > - unsigned int reg; > > > > regcache_cache_only(es8326->regmap, false); > > regcache_sync(es8326->regmap); > > > > + /* reset register value to default */ > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); > > + usleep_range(1000, 3000); > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); > > This looks wrong, you're resyncing the cache and then start resetting > registers? It feels like the ordering is off here, and some of this > reset sequence might want to be done with the cache bypassed. Are you > sure that there's no corruption of user visible state resulting from the > power up sequence, especially around the HP driver? Basically the chip runs through the start-up sequence every time in the resume function. Will move the reset sequence to es8326_suspend

2 1

[PATCH v2 0/5] Few audio fixes on Tegra platforms
by Sameer Pujar 13 Jul '23

13 Jul '23

This series fixes some of the issues which were observed during an attempt to enhance automated test coverage on Jetson AGX Orin. Below is a short summary of the issues and fixes: * Sample rate coversion failures above 48kHz. * AMX and ADX test cases failures due to incorrect byte mask. * Atomic sleep in RT5640 codec which is present on Jetson AGX Orin. * AHUB clock fixes on Tegra234 and previous chips. * Minor cleanups in ASRC and AHUB driver. Changelog ========= v1 -> v2: --------- * Few patches got accepted in the original (v1) series. Now v2 addresses comments for remaining patches. * AMX/ADX byte map fix patch is updated with more details in the commit message and added TODO item in the driver to improve the logic. * For RT5640 codec patch, the threaded IRQ is used for only for rt5640_irq() and rt5640_jd_gpio_irq() is left untouched. Sameer Pujar (2): ASoC: rt5640: Fix sleep in atomic context arm64: tegra: Update AHUB clock parent and rate Sheetal (3): ASoC: tegra: Fix AMX byte map ASoC: tegra: Fix ADX byte map arm64: tegra: Update AHUB clock parent and rate on Tegra234 arch/arm64/boot/dts/nvidia/tegra186.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra210.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra234.dtsi | 3 ++- sound/soc/codecs/rt5640.c | 12 ++++++---- sound/soc/tegra/tegra210_adx.c | 34 +++++++++++++++++---------- sound/soc/tegra/tegra210_amx.c | 40 ++++++++++++++++++-------------- 7 files changed, 59 insertions(+), 39 deletions(-) -- 2.7.4

4 11

[PATCH 1/3] ALSA: emu10k1: fix return value of snd_emu1010_adc_pads_put()
by Oswald Buddenhagen 13 Jul '23

13 Jul '23

It returned zero even if the value had changed. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- this should have been in cc766807a2 (fix return value of snd_emu1010_dac_pads_put()), but, well. --- sound/pci/emu10k1/emumixer.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index f9500cd50a4b..573e1c7c5e50 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -770,18 +770,21 @@ static int snd_emu1010_adc_pads_put(struct snd_kcontrol *kcontrol, struct snd_ct struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); unsigned int mask = snd_emu1010_adc_pad_regs[kcontrol->private_value]; unsigned int val, cache; + int change; + val = ucontrol->value.integer.value[0]; cache = emu->emu1010.adc_pads; if (val == 1) cache = cache | mask; else cache = cache & ~mask; - if (cache != emu->emu1010.adc_pads) { + change = (cache != emu->emu1010.adc_pads); + if (change) { snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, cache ); emu->emu1010.adc_pads = cache; } - return 0; + return change; } static const struct snd_kcontrol_new emu1010_adc_pads_ctl = { -- 2.40.0.152.g15d061e6df

2 14

[PATCH] ALSA: pcmtest: Extend error injection, decrease buffer filling overhead
by Ivan Orlov 13 Jul '23

13 Jul '23

Extend 'pcmtest' virtual driver with 'open' callback error injection functionality, as it already can inject errors into other PCM callbacks. Fix the driver to use already defined variables where it is possible. Additionally, decrease the buffer filling overhead with conditional reminder calculation in the 'inc_buf_pos' inline function. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- sound/drivers/pcmtest.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 291e7fe47893..08e14b5eb772 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -65,6 +65,7 @@ static int inject_delay; static bool inject_hwpars_err; static bool inject_prepare_err; static bool inject_trigger_err; +static bool inject_open_err; static short fill_mode = FILL_MODE_PAT; @@ -88,6 +89,9 @@ module_param(inject_prepare_err, bool, 0600); MODULE_PARM_DESC(inject_prepare_err, "Inject EINVAL error in the 'prepare' callback"); module_param(inject_trigger_err, bool, 0600); MODULE_PARM_DESC(inject_trigger_err, "Inject EINVAL error in the 'trigger' callback"); +module_param(inject_open_err, bool, 0600); +MODULE_PARM_DESC(inject_open_err, "Inject EBUSY error in the 'open' callback"); + struct pcmtst { struct snd_pcm *pcm; @@ -140,7 +144,8 @@ static inline void inc_buf_pos(struct pcmtst_buf_iter *v_iter, size_t by, size_t { v_iter->total_bytes += by; v_iter->buf_pos += by; - v_iter->buf_pos %= bytes; + if (v_iter->buf_pos >= bytes) + v_iter->buf_pos %= bytes; } /* @@ -196,10 +201,10 @@ static void check_buf_block_ni(struct pcmtst_buf_iter *v_iter, struct snd_pcm_ru u8 current_byte; for (i = 0; i < v_iter->b_rw; i++) { - current_byte = runtime->dma_area[buf_pos_n(v_iter, channels, i % channels)]; + ch_num = i % channels; + current_byte = runtime->dma_area[buf_pos_n(v_iter, channels, ch_num)]; if (!current_byte) break; - ch_num = i % channels; if (current_byte != patt_bufs[ch_num].buf[(v_iter->total_bytes / channels) % patt_bufs[ch_num].len]) { v_iter->is_buf_corrupted = true; @@ -239,7 +244,7 @@ static void fill_block_pattern_n(struct pcmtst_buf_iter *v_iter, struct snd_pcm_ for (i = 0; i < v_iter->b_rw; i++) { ch_num = i % channels; - runtime->dma_area[buf_pos_n(v_iter, channels, i % channels)] = + runtime->dma_area[buf_pos_n(v_iter, channels, ch_num)] = patt_bufs[ch_num].buf[(v_iter->total_bytes / channels) % patt_bufs[ch_num].len]; inc_buf_pos(v_iter, 1, runtime->dma_bytes); @@ -364,6 +369,9 @@ static int snd_pcmtst_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct pcmtst_buf_iter *v_iter; + if (inject_open_err) + return -EBUSY; + v_iter = kzalloc(sizeof(*v_iter), GFP_KERNEL); if (!v_iter) return -ENOMEM; -- 2.34.1

2 2

[PATCH][next] selftests: ALSA: Fix fclose on an already fclosed file pointer
by Colin Ian King 13 Jul '23

13 Jul '23

In the case where a sysfs file cannot be opened the error return path fcloses file pointer fpl, however, fpl has already been closed in the previous stanza. Fix the double fclose by removing it. Fixes: 10b98a4db11a ("selftests: ALSA: Add test for the 'pcmtest' driver") Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- tools/testing/selftests/alsa/test-pcmtest-driver.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/tools/testing/selftests/alsa/test-pcmtest-driver.c b/tools/testing/selftests/alsa/test-pcmtest-driver.c index 71931b240a83..357adc722cba 100644 --- a/tools/testing/selftests/alsa/test-pcmtest-driver.c +++ b/tools/testing/selftests/alsa/test-pcmtest-driver.c @@ -47,10 +47,8 @@ static int read_patterns(void) sprintf(pf, "/sys/kernel/debug/pcmtest/fill_pattern%d", i); fp = fopen(pf, "r"); - if (!fp) { - fclose(fpl); + if (!fp) return -1; - } fread(patterns[i].buf, 1, patterns[i].len, fp); fclose(fp); } -- 2.39.2

3 2

[PATCH v4] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 12 Jul '23

12 Jul '23

This configuration supports JSL boards which implement ALC5650 dual I2S interface codec. Two DAI links are added: AIF1 (on codec side) for headphone and AIF2 for speakers. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/sof_rt5682.c | 80 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 3 files changed, 93 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f472f603ab75..1fe830af2b84 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -475,7 +475,7 @@ endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC if SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_SOF_RT5682_MACH - tristate "SOF with rt5682 codec in I2S Mode" + tristate "SOF with rt5650/rt5682 codec in I2S Mode" depends on I2C && ACPI depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ @@ -485,6 +485,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P + select SND_SOC_RT5645 select SND_SOC_RT5682_I2C select SND_SOC_RT5682S select SND_SOC_DMIC @@ -494,7 +495,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_INTEL_SOF_REALTEK_COMMON help This adds support for ASoC machine driver for SOF platforms - with rt5682 codec. + with rt5650 or rt5682 codec. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7c034d671cf3..b4f07bdcf8b4 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -22,6 +22,7 @@ #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" +#include "../../codecs/rt5645.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" @@ -60,6 +61,7 @@ #define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) #define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) #define SOF_RT1019_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_RT5650_HEADPHONE_CODEC_PRESENT BIT(27) /* Default: MCLK on, MCLK 19.2M, SSP0 */ @@ -305,6 +307,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; + int extra_jack_data; int ret; /* need to enable ASRC function for 24MHz mclk rate */ @@ -315,7 +318,16 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) RT5682S_DA_STEREO1_FILTER | RT5682S_AD_STEREO1_FILTER, RT5682S_CLK_SEL_I2S1_ASRC); - else + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(component, + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | RT5682_AD_STEREO1_FILTER, @@ -365,7 +377,12 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); + + if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + extra_jack_data = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra_jack_data); + } else + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); @@ -402,6 +419,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_MCLK; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_MCLK; else pll_source = RT5682_PLL1_S_MCLK; @@ -422,6 +441,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, } else { if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_BCLK1; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_BCLK1; else pll_source = RT5682_PLL1_S_BCLK1; @@ -431,6 +452,9 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + pll_id = 0; /* not used in codec driver */ + clk_id = RT5645_SCLK_S_PLL1; } else { pll_id = RT5682_PLL1; clk_id = RT5682_SCLK_S_PLL1; @@ -559,11 +583,30 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "SPOL" }, + { "Right Spk", NULL, "SPOR" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes, + ARRAY_SIZE(rt5650_spk_dapm_routes)); + if (ret) + dev_err(rtd->dev, "fail to add dapm routes, ret=%d\n", ret); + + return ret; +} + static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -614,6 +657,17 @@ static struct snd_soc_dai_link_component rt5682s_component[] = { } }; +static struct snd_soc_dai_link_component rt5650_components[] = { + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif1", + }, + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif2", + } +}; + static struct snd_soc_dai_link_component dmic_component[] = { { .name = "dmic-codec", @@ -652,6 +706,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { links[id].codecs = rt5682s_component; links[id].num_codecs = ARRAY_SIZE(rt5682s_component); + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[0]; + links[id].num_codecs = 1; } else { links[id].codecs = rt5682_component; links[id].num_codecs = ARRAY_SIZE(rt5682_component); @@ -804,6 +861,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].init = max_98390_spk_codec_init; links[id].ops = &max_98390_ops; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[1]; + links[id].num_codecs = 1; + links[id].init = rt5650_spk_init; + links[id].ops = &sof_rt5682_ops; } else { max_98357a_dai_link(&links[id]); } @@ -890,6 +952,12 @@ static int sof_audio_probe(struct platform_device *pdev) /* Detect the headset codec variant */ if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + else if (acpi_dev_present("10EC5650", NULL, -1)) { + sof_rt5682_quirk |= SOF_RT5650_HEADPHONE_CODEC_PRESENT; + + sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", + GFP_KERNEL); + } if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; @@ -1178,6 +1246,14 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(0) | SOF_RT5682_NUM_HDMIDEV(3)), }, + { + .name = "jsl_rt5650", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index f5c7e1bbded0..f56bd7d656e9 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -34,6 +34,11 @@ static const struct snd_soc_acpi_codecs mx98360a_spk = { .codecs = {"MX98360A"} }; +static struct snd_soc_acpi_codecs rt5650_spk = { + .num_codecs = 1, + .codecs = {"10EC5650"} +}; + static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {"10EC5682", "RTL5682"}, @@ -98,6 +103,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .id = "10EC5650", + .drv_name = "jsl_rt5650", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt5650_spk, + .sof_tplg_filename = "sof-jsl-rt5650.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

3 2

[PATCH 00/13] PCI: Define Intel PCI IDs and use them in drivers
by Amadeusz Sławiński 12 Jul '23

12 Jul '23

PCI IDs for Intel HDA are duplicated across quite a few drivers, due to various configurations and historical reasons. Currently almost all uses of HDA PCI IDs have corresponding comment telling which platform it is. Additionally there are some inconsistencies between drivers about which ID corresponds to which device. Simplify things, by adding PCI IDs to global header and make use of them in drivers. This allows for removal of comments by having IDs themselves being self explanatory. Additionally it allows for removal of existing inconsistencies by having one source of truth. Changes from RFC: - Sort Intel PCI IDs before adding new ones - Fix ordering of new PCI IDs (Andy) - Define all used Intel IDs (Andy) - Add macros for controller type detection (Andy/Bjorn) - Add set of patches changing to use above macro (Andy/Bjorn) - Use PCI_DEVICE_DATA for Intel IDs in sound/pci/hda/hda_intel.c (Andy) - Commit message wording (Andy) - Remove unnecessary tabs (Andy) Amadeusz Sławiński (13): PCI: Sort Intel PCI IDs by number PCI: Add Intel Audio DSP devices to pci_ids.h ALSA: hda: Add controller matching macros ALSA: hda: Use global PCI match macro ALSA: hda/i915: Use global PCI match macro ASoC: Intel: Skylake: Use global PCI match macro ALSA: intel-dsp-config: Convert to PCI device IDs defines ALSA: hda: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: Skylake: Convert to PCI device IDs defines ASoC: SOF: Intel: Convert to PCI device IDs defines ASoC: Intel: sst: Convert to PCI device IDs defines include/linux/pci_ids.h | 104 +++++-- include/sound/hda_codec.h | 3 - include/sound/hdaudio.h | 27 ++ sound/hda/hdac_i915.c | 7 +- sound/hda/intel-dsp-config.c | 119 ++++---- sound/pci/hda/hda_intel.c | 373 ++++++++++--------------- sound/soc/intel/atom/sst/sst.c | 3 +- sound/soc/intel/atom/sst/sst.h | 1 - sound/soc/intel/atom/sst/sst_pci.c | 4 +- sound/soc/intel/avs/board_selection.c | 10 +- sound/soc/intel/avs/core.c | 16 +- sound/soc/intel/skylake/skl-messages.c | 16 +- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/intel/skylake/skl.c | 36 +-- sound/soc/sof/intel/pci-apl.c | 9 +- sound/soc/sof/intel/pci-cnl.c | 15 +- sound/soc/sof/intel/pci-icl.c | 12 +- sound/soc/sof/intel/pci-mtl.c | 3 +- sound/soc/sof/intel/pci-skl.c | 6 +- sound/soc/sof/intel/pci-tgl.c | 45 +-- sound/soc/sof/intel/pci-tng.c | 3 +- 21 files changed, 384 insertions(+), 431 deletions(-) -- 2.34.1

7 42

[PATCH v2 0/4] ASoC: rt5677: Refactor GPIO and use device_get_match_data()
by Andy Shevchenko 12 Jul '23

12 Jul '23

The code can be simplified with refactored GPIO parts and with use of device_get_match_data(). Besides that couple of additional changes, one for maintenance and one for making IRQ domain agnostic (not being pinned to OF). Changelog v2: - refactored GPIO code in (a new) patch 1 - fixed compilation error in patch 2 (LKP) Andy Shevchenko (4): ASoC: rt5677: Refactor GPIO support code ASoC: rt5677: Use agnostic irq_domain_create_linear() ASoC: rt5677: Use device_get_match_data() ASoC: rt5677: Sort headers alphabetically sound/soc/codecs/rt5677.c | 117 +++++++++++++------------------------- sound/soc/codecs/rt5677.h | 92 +++++------------------------- 2 files changed, 53 insertions(+), 156 deletions(-) -- 2.40.0.1.gaa8946217a0b

2 8

[PATCH] ASoC: tlv320aic32x4: Fix the divide by zero
by Guiting Shen 12 Jul '23

12 Jul '23

The value of register(NDAC,MDAC,NADC,MADC,BCLKN) maybe zero lead to divide by zero in clk_aic32x4_div_recalc_rate().And the rate should be divide by 128 if the value was zero in this function according to the datasheet. Add the macro AIC32X4_DIV_MAX to present the 128 and return 0 if failing to read the value of register. Signed-off-by: Guiting Shen <aarongt.shen(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4-clk.c | 16 +++++++++++----- sound/soc/codecs/tlv320aic32x4.h | 5 +++-- 2 files changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index c116e82f712d..5c0a76a4a106 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -321,7 +321,7 @@ static int clk_aic32x4_div_set_rate(struct clk_hw *hw, unsigned long rate, u8 divisor; divisor = DIV_ROUND_UP(parent_rate, rate); - if (divisor > 128) + if (divisor > AIC32X4_DIV_MAX) return -EINVAL; return regmap_update_bits(div->regmap, div->reg, @@ -334,7 +334,7 @@ static int clk_aic32x4_div_determine_rate(struct clk_hw *hw, unsigned long divisor; divisor = DIV_ROUND_UP(req->best_parent_rate, req->rate); - if (divisor > 128) + if (divisor > AIC32X4_DIV_MAX) return -EINVAL; req->rate = DIV_ROUND_UP(req->best_parent_rate, divisor); @@ -345,12 +345,18 @@ static unsigned long clk_aic32x4_div_recalc_rate(struct clk_hw *hw, unsigned long parent_rate) { struct clk_aic32x4 *div = to_clk_aic32x4(hw); - unsigned int val; + int err; + + err = regmap_read(div->regmap, div->reg, &val); + if (err) + return 0; - regmap_read(div->regmap, div->reg, &val); + val &= AIC32X4_DIV_MASK; + if (!val) + val = AIC32X4_DIV_MAX; - return DIV_ROUND_UP(parent_rate, val & AIC32X4_DIV_MASK); + return DIV_ROUND_UP(parent_rate, val); } static const struct clk_ops aic32x4_div_ops = { diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index 4de5bd9e8cc5..d6101ce73f80 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -223,8 +223,9 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name); #define AIC32X4_REFPOWERUP_120MS 0x07 /* Common mask and enable for all of the dividers */ -#define AIC32X4_DIVEN BIT(7) -#define AIC32X4_DIV_MASK GENMASK(6, 0) +#define AIC32X4_DIVEN BIT(7) +#define AIC32X4_DIV_MASK GENMASK(6, 0) +#define AIC32X4_DIV_MAX 128 /* Clock Limits */ #define AIC32X4_MAX_DOSR_FREQ 6200000 -- 2.25.1

1 0

[PATCH v2 00/37] ALSA: Add MIDI 2.0 support
by Takashi Iwai 12 Jul '23

12 Jul '23

Hi, this is a revised patch set for adding the support of MIDI 2.0 functionality. Only a few minor updates since v1 as below: * Bump CTL protocol version to 2.0.9 * Add two more ioctls for control API to receive UMP EP and block info * Minor compile warning fixes * Documentation fix and addition of two new ioctls * Put acks The latest code is found in topic/midi20 branch of sound git tree. And below is the cover letter of v1, which is still valid: Hi, this is a (largish) patch set for adding the support of MIDI 2.0 functionality, mainly targeted for USB devices. MIDI 2.0 is a complete overhaul of the 40-years old MIDI 1.0. Unlike MIDI 1.0 byte stream, MIDI 2.0 uses packets in 32bit words for Universal MIDI Packet (UMP) protocol. It supports both MIDI 1.0 commands for compatibility and the extended MIDI 2.0 commands for higher resolutions and more functions. For supporting the UMP, the patch set extends the existing ALSA rawmidi and sequencer interfaces, and adds the USB MIDI 2.0 support to the standard USB-audio driver. The rawmidi for UMP has a different device name (/dev/snd/umpC*D*) and it reads/writes UMP packet data in 32bit CPU-native endianness. For the old MIDI 1.0 applications, the legacy rawmidi interface is provided, too. As default, USB-audio driver will take the alternate setting for MIDI 2.0 interface, and the compatibility with MIDI 1.0 is provided via the rawmidi common layer. However, user may let the driver falling back to the old MIDI 1.0 interface by a module option, too. A UMP-capable rawmidi device can create the corresponding ALSA sequencer client(s) to support the UMP Endpoint and UMP Group connections. As a nature of ALSA sequencer, arbitrary connections between clients/ports are allowed, and the ALSA sequencer core performs the automatic conversions for the connections between a new UMP sequencer client and a legacy MIDI 1.0 sequencer client. It allows the existing application to use MIDI 2.0 devices without changes. The MIDI-CI, which is another major extension in MIDI 2.0, isn't covered by this patch set. It would be implemented rather in user-space. Roughly speaking, the first half of this patch set is for extending the rawmidi and USB-audio, and the second half is for extending the ALSA sequencer interface. The patch set is based on 6.4-rc2 kernel, but all patches can be cleanly applicable on 6.2 and 6.3 kernels, too (while 6.1 and older kernels would need minor adjustment for uapi header changes). The updates for alsa-lib and alsa-utils will follow shortly later. The author thanks members of MIDI Association OS/API Working Group, especially Andrew Mee, for great helps for the initial design and debugging / testing the drivers. Takashi --- Takashi Iwai (37): ALSA: rawmidi: Pass rawmidi directly to snd_rawmidi_kernel_open() ALSA: rawmidi: Add ioctl callback to snd_rawmidi_global_ops ALSA: rawmidi: UMP support ALSA: rawmidi: Skip UMP devices at SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE ALSA: ump: Add ioctls to inquiry UMP EP and Block info via control API ALSA: ump: Additional proc output ALSA: usb-audio: Manage number of rawmidis globally ALSA: usb-audio: Define USB MIDI 2.0 specs ALSA: usb-audio: USB MIDI 2.0 UMP support ALSA: usb-audio: Get UMP EP name string from USB interface ALSA: usb-audio: Trim superfluous "MIDI" suffix from UMP EP name ALSA: usb-audio: Create UMP blocks from USB MIDI GTBs ALSA: ump: Redirect rawmidi substream access via own helpers ALSA: ump: Add legacy raw MIDI support ALSA: usb-audio: Enable the legacy raw MIDI support ALSA: usb-audio: Inform inconsistent protocols in GTBs ALSA: seq: Clear padded bytes at expanding events ALSA: seq: Add snd_seq_expand_var_event_at() helper ALSA: seq: Treat snd_seq_client object directly in client drivers ALSA: seq: Drop dead code for the old broadcast support ALSA: seq: Check the conflicting port at port creation ALSA: seq: Check validity before creating a port object ALSA: seq: Prohibit creating ports with special numbers ALSA: seq: Introduce SNDRV_SEQ_IOCTL_USER_PVERSION ioctl ALSA: seq: Add UMP support ALSA: seq: Add port inactive flag ALSA: seq: Support MIDI 2.0 UMP Endpoint port ALSA: seq: Add port direction to snd_seq_port_info ALSA: seq: Add UMP group number to snd_seq_port_info ALSA: seq: Automatic conversion of UMP events ALSA: seq: Allow suppressing UMP conversions ALSA: seq: Bind UMP device ALSA: seq: ump: Create UMP Endpoint port for broadcast ALSA: seq: Add ioctls for client UMP info query and setup ALSA: seq: Print UMP Endpoint and Block information in proc outputs ALSA: seq: Add UMP group filter ALSA: docs: Add MIDI 2.0 documentation Documentation/sound/designs/index.rst | 1 + Documentation/sound/designs/midi-2.0.rst | 351 +++++++ include/linux/usb/midi-v2.h | 94 ++ include/sound/asequencer.h | 4 + include/sound/rawmidi.h | 15 +- include/sound/seq_device.h | 1 + include/sound/seq_kernel.h | 10 + include/sound/ump.h | 175 ++++ include/sound/ump_msg.h | 540 ++++++++++ include/uapi/sound/asequencer.h | 83 +- include/uapi/sound/asound.h | 62 +- sound/core/Kconfig | 13 + sound/core/Makefile | 3 + sound/core/rawmidi.c | 264 +++-- sound/core/rawmidi_compat.c | 4 + sound/core/seq/Kconfig | 14 + sound/core/seq/Makefile | 3 + sound/core/seq/seq_clientmgr.c | 577 +++++++---- sound/core/seq/seq_clientmgr.h | 27 +- sound/core/seq/seq_compat.c | 3 + sound/core/seq/seq_dummy.c | 9 + sound/core/seq/seq_memory.c | 98 +- sound/core/seq/seq_memory.h | 19 +- sound/core/seq/seq_midi.c | 12 +- sound/core/seq/seq_ports.c | 47 +- sound/core/seq/seq_ports.h | 23 +- sound/core/seq/seq_ump_client.c | 464 +++++++++ sound/core/seq/seq_ump_convert.c | 1203 ++++++++++++++++++++++ sound/core/seq/seq_ump_convert.h | 22 + sound/core/seq/seq_virmidi.c | 1 + sound/core/ump.c | 668 ++++++++++++ sound/core/ump_convert.c | 520 ++++++++++ sound/core/ump_convert.h | 43 + sound/usb/Kconfig | 11 + sound/usb/Makefile | 1 + sound/usb/card.c | 12 +- sound/usb/midi.c | 7 +- sound/usb/midi.h | 5 +- sound/usb/midi2.c | 1190 +++++++++++++++++++++ sound/usb/midi2.h | 33 + sound/usb/quirks.c | 8 +- sound/usb/usbaudio.h | 2 + 42 files changed, 6306 insertions(+), 336 deletions(-) create mode 100644 Documentation/sound/designs/midi-2.0.rst create mode 100644 include/linux/usb/midi-v2.h create mode 100644 include/sound/ump.h create mode 100644 include/sound/ump_msg.h create mode 100644 sound/core/seq/seq_ump_client.c create mode 100644 sound/core/seq/seq_ump_convert.c create mode 100644 sound/core/seq/seq_ump_convert.h create mode 100644 sound/core/ump.c create mode 100644 sound/core/ump_convert.c create mode 100644 sound/core/ump_convert.h create mode 100644 sound/usb/midi2.c create mode 100644 sound/usb/midi2.h -- 2.35.3

4 68

ALC4082 - ASRock Z790 PG-ITX/TB4
by GitHub issues - edited 12 Jul '23

12 Jul '23

alsa-project/alsa-ucm-conf issue #334 was edited from Shmob1: I have an [ASRock Z790 PG-ITX/TB4](https://pg.asrock.com/MB/Intel/Z790%20PG-ITXTB4/index.asp) which has the `ALC4082` Realtek chipset connected over USB, that has been mentioned in a number of issues, most recently #229. It seems the configuration changes in that issue (notably, https://github.com/hwchong/alsa-ucm-conf/commit/3ec7e6e2a61240764b77c30892f…) work on my board as well, with a small modification. Unlike the Taichi boards, my Realtek audio driver is at `USB26ce:0a08`, not `USB26ce:0a06`. Making the necessary regex changes in `ALC4080-HiFi.conf` and `USB-Audio.conf` enables functionality on my board. I am happy to make a PR to merge these changes if pertinent. Thanks for the great work! Update: I am unable to get more than 2 channels through spdif either using AC3 or DTS Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/334 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND 0/2] ASoC: mt8188: add memory-region support
by Trevor Wu 12 Jul '23

12 Jul '23

This series introduces support for memory-region, allowing afe memif to utilize the region specified in dts. Trevor Wu (2): ASoC: mediatek: mt8188: add memory-region support ASoC: dt-bindings: mediatek,mt8188-afe: add memory-region .../devicetree/bindings/sound/mediatek,mt8188-afe.yaml | 7 +++++++ sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 8 ++++++-- 2 files changed, 13 insertions(+), 2 deletions(-) -- 2.18.0

2 4

[PATCH 1/4] ASoC: codecs: es8326: Add es8326_mute function
by Zhu Ning 12 Jul '23

12 Jul '23

The internal analog power and hp Vref of es8326 should always be on to reduce pop noise. The HP_VOL and HP_CAL are moved to es8326_mute function so they are turned on at last and turned off at first. Also, the calibration should be done manually once during start-up to reduce DC offset on headphone. Signed-off-by: Zhu Ning <zhuning0077(a)gmail.com> --- sound/soc/codecs/es8326.c | 75 ++++++++++++++++----------------------- sound/soc/codecs/es8326.h | 5 +-- 2 files changed, 34 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index ed348bbf5f49..9190a8779334 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -38,6 +38,9 @@ struct es8326_priv { u8 interrupt_clk; bool jd_inverted; unsigned int sysclk; + + bool calibrated; + int version; }; static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); @@ -121,33 +124,12 @@ static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { /* Analog Power Supply*/ SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN, 0, 1), SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN, 1, 1), - SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN, 7, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN, 6, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN, 5, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN, 4, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN, 3, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX, 3, 0, NULL, 0), - /* Headphone Charge Pump and Output */ - SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL, 7, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL, 3, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER, - 3, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER, - 2, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER, - 1, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER, - 0, 1, NULL, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL, - ES8326_HPOR_SHIFT, 7, 7, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL, - 0, 7, 7, 0), - SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), }; @@ -166,34 +148,12 @@ static const struct snd_soc_dapm_route es8326_dapm_routes[] = { {"I2S OUT", NULL, "ADC L"}, {"I2S OUT", NULL, "ADC R"}, - {"I2S OUT", NULL, "Analog Power"}, - {"I2S OUT", NULL, "ADC Vref"}, - {"I2S OUT", NULL, "Vref Power"}, - {"I2S OUT", NULL, "IBias Power"}, - {"I2S IN", NULL, "Analog Power"}, - {"I2S IN", NULL, "DAC Vref"}, - {"I2S IN", NULL, "Vref Power"}, - {"I2S IN", NULL, "IBias Power"}, - {"Right DAC", NULL, "I2S IN"}, {"Left DAC", NULL, "I2S IN"}, {"LHPMIX", NULL, "Left DAC"}, {"RHPMIX", NULL, "Right DAC"}, - {"HPOR", NULL, "HPOR Cal"}, - {"HPOL", NULL, "HPOL Cal"}, - {"HPOR", NULL, "HPOR Supply"}, - {"HPOL", NULL, "HPOL Supply"}, - {"HPOL", NULL, "Headphone Charge Pump"}, - {"HPOR", NULL, "Headphone Charge Pump"}, - {"HPOL", NULL, "Headphone Driver Bias"}, - {"HPOR", NULL, "Headphone Driver Bias"}, - {"HPOL", NULL, "Headphone LDO"}, - {"HPOR", NULL, "Headphone LDO"}, - {"HPOL", NULL, "Headphone Reference"}, - {"HPOR", NULL, "Headphone Reference"}, - {"HPOL", NULL, "LHPMIX"}, {"HPOR", NULL, "RHPMIX"}, }; @@ -419,6 +379,31 @@ static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } +static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) +{ + struct snd_soc_component *component = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + if (mute) { + regmap_write(es8326->regmap, ES8326_HP_CAL, 0x00); + regmap_update_bits(es8326->regmap, ES8326_DAC_MUTE, + ES8326_MUTE_MASK, ES8326_MUTE); + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xf0); + } else { + if (!es8326->calibrated) { + regmap_write(es8326->regmap, ES8326_HP_CAL, 0x88); + msleep(30); + es8326->calibrated = true; + } + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xa0); + regmap_write(es8326->regmap, ES8326_HP_VOL, 0x00); + regmap_write(es8326->regmap, ES8326_HP_CAL, 0x77); + regmap_update_bits(es8326->regmap, ES8326_DAC_MUTE, + ES8326_MUTE_MASK, ~(ES8326_MUTE)); + } + return 0; +} + static int es8326_set_bias_level(struct snd_soc_component *codec, enum snd_soc_bias_level level) { @@ -469,6 +454,8 @@ static const struct snd_soc_dai_ops es8326_ops = { .hw_params = es8326_pcm_hw_params, .set_fmt = es8326_set_dai_fmt, .set_sysclk = es8326_set_dai_sysclk, + .mute_stream = es8326_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver es8326_dai = { @@ -691,7 +678,7 @@ static int es8326_suspend(struct snd_soc_component *component) cancel_delayed_work_sync(&es8326->jack_detect_work); es8326_disable_micbias(component); - + es8326->calibrated = false; regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); regcache_cache_only(es8326->regmap, true); regcache_mark_dirty(es8326->regmap); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h index 8e5ffe5ee10d..b837ed8c4b39 100644 --- a/sound/soc/codecs/es8326.h +++ b/sound/soc/codecs/es8326.h @@ -9,8 +9,6 @@ #ifndef _ES8326_H #define _ES8326_H -#define CONFIG_HHTECH_MINIPMP 1 - /* ES8326 register space */ #define ES8326_RESET 0x00 #define ES8326_CLK_CTL 0x01 @@ -94,6 +92,8 @@ #define ES8326_PWRUP_SEQ_EN (1 << 5) #define ES8326_CODEC_RESET (0x0f << 0) #define ES8326_CSM_OFF (0 << 7) +#define ES8326_MUTE_MASK (3 << 0) +#define ES8326_MUTE (3 << 0) /* ES8326_CLK_CTL */ #define ES8326_CLK_ON (0x7f << 0) @@ -180,3 +180,4 @@ #define ES8326_VERSION_B (1 << 0) #endif + -- 2.34.1

2 4

[PATCH 0/3] regmap: Fix REGMAP selections
by Geert Uytterhoeven 12 Jul '23

12 Jul '23

Hi all, This patch series fixes several missing selects related to REGMAP. The first patch fixes a serious thinko I made when making the regmap KUnit test depend on REGMAP. The last two patches fix missing selects I discovered when diving deeper into the REGMAP_* => REGMAP logic. Thanks for your comments! Geert Uytterhoeven (3): regmap: Replace "default y if" by select regmap: REGMAP_SLIMBUS should select REGMAP ASoC: codecs: SND_SOC_WCD934X should select REGMAP_IRQ drivers/base/regmap/Kconfig | 16 +++++++++++++++- drivers/mfd/Kconfig | 1 - sound/soc/codecs/Kconfig | 1 + 3 files changed, 16 insertions(+), 2 deletions(-) -- 2.34.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

3 11

[PATCH v2 00/15] spi: Header and core clean up and refactoring
by Andy Shevchenko 12 Jul '23

12 Jul '23

Various cleanups and refactorings of the SPI header and core parts united in a single series. It also touches drivers under SPI subsystem folder on the pure renaming purposes of some constants. No functional change intended. Changelog v2: - added new patches 3,4,5,10,13,14 - massaged comment and kernel doc in patch 9 - split used to be patch 4 to patches 11,12 - covered a few things in SPI core in patch 15 - amended commit message for above (Mark) - reshuffled patches in the series for better logical grouping Andy Shevchenko (15): spi: Remove unneeded OF node NULL checks spi: Drop duplicate IDR allocation code in spi_register_controller() spi: Replace if-else-if by bitops and multiplications spi: Replace open coded spi_controller_xfer_timeout() spi: Remove code duplication in spi_add_device_locked() spi: Use sysfs_emit() to instead of s*printf() spi: Sort headers alphabetically spi: Clean up headers spi: Use struct_size() helper spi: Use predefined constants from bits.h and units.h spi: Get rid of old SPI_MASTER_NO_TX & SPI_MASTER_NO_RX spi: Get rid of old SPI_MASTER_MUST_TX & SPI_MASTER_MUST_RX spi: Rename SPI_MASTER_GPIO_SS to SPI_CONTROLLER_GPIO_SS spi: Convert to SPI_CONTROLLER_HALF_DUPLEX spi: Fix spelling typos and acronyms capitalization drivers/spi/spi-amd.c | 2 +- drivers/spi/spi-at91-usart.c | 2 +- drivers/spi/spi-ath79.c | 2 +- drivers/spi/spi-atmel.c | 4 +- drivers/spi/spi-bitbang-txrx.h | 16 +-- drivers/spi/spi-bitbang.c | 8 +- drivers/spi/spi-cavium-thunderx.c | 2 +- drivers/spi/spi-davinci.c | 2 +- drivers/spi/spi-dw-core.c | 2 +- drivers/spi/spi-falcon.c | 2 +- drivers/spi/spi-fsl-lpspi.c | 2 +- drivers/spi/spi-gpio.c | 10 +- drivers/spi/spi-imx.c | 2 +- drivers/spi/spi-lp8841-rtc.c | 10 +- drivers/spi/spi-meson-spicc.c | 2 +- drivers/spi/spi-mt65xx.c | 2 +- drivers/spi/spi-mxs.c | 2 +- drivers/spi/spi-omap-uwire.c | 2 +- drivers/spi/spi-orion.c | 2 +- drivers/spi/spi-pci1xxxx.c | 2 +- drivers/spi/spi-pic32-sqi.c | 2 +- drivers/spi/spi-pic32.c | 2 +- drivers/spi/spi-qcom-qspi.c | 2 +- drivers/spi/spi-rb4xx.c | 2 +- drivers/spi/spi-rockchip-sfc.c | 2 +- drivers/spi/spi-rockchip.c | 2 +- drivers/spi/spi-sifive.c | 2 +- drivers/spi/spi-slave-mt27xx.c | 2 +- drivers/spi/spi-sprd-adi.c | 2 +- drivers/spi/spi-stm32.c | 2 +- drivers/spi/spi-ti-qspi.c | 2 +- drivers/spi/spi-xcomm.c | 2 +- drivers/spi/spi-xtensa-xtfpga.c | 2 +- drivers/spi/spi.c | 204 ++++++++++++------------------ include/linux/spi/spi.h | 198 +++++++++++++++++------------ include/trace/events/spi.h | 2 +- 36 files changed, 247 insertions(+), 261 deletions(-) -- 2.40.0.1.gaa8946217a0b

6 47

[PATCH 00/14] ASoC: add multi Component support
by Kuninori Morimoto 12 Jul '23

12 Jul '23

Hi Mark On below HW case, we would like to use it as "2 Cards", but unfortunately it is impossible in intuitive way or possible but not intuitive way. In reality, it is handled as "1 big Card" today. +-- basic board --------+ |+--------+ | || CPU ch0| <--> CodecA | || ch1| <-+ | |+--------+ | | +-------------|---------+ +-- expansion board ----+ | | | | +-> CodecB| +-----------------------+ To handling it as intuitive "2 Cards", this patch-set adds multi Component support. Link: https://lore.kernel.org/all/87a6b6cofh.wl-kuninori.morimoto.gx@renesas.com/ Kuninori Morimoto (14): ASoC: soc-core.c: initialize dlc on snd_soc_get_dai_id() ASoC: soc-core.c: cleanup soc_dai_link_sanity_check() ASoC: soc-dai.c: add DAI get/match functions ASoC: soc-core.c: enable multi Component ASoC: soc-core.c: add snd_soc_get_dai_via_args() ASoC: soc-core.c: add snd_soc_dlc_use_cpu_as_platform() ASoC: soc-core.c: add snd_soc_copy_dai_args() ASoC: simple-card-utils.c: enable multi Component support ASoC: simple-card.c: enable multi Component support ASoC: rsnd: use DAI driver ID instead of DAI ID ASoC: rsnd: cleanup rsnd_dai_of_node() ASoC: rsnd: enable multi Component support for Audio Graph Card/Card2 ASoC: dt-bindings: renesas,rsnd.yaml: add common port-def ASoC: dt-bindings: renesas,rsnd.yaml: enable multi ports for multi Component support .../bindings/sound/renesas,rsnd.yaml | 60 ++-- include/sound/simple_card_utils.h | 5 +- include/sound/soc-dai.h | 3 + include/sound/soc.h | 6 + sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/generic/simple-card-utils.c | 23 +- sound/soc/generic/simple-card.c | 20 +- sound/soc/sh/rcar/core.c | 146 +++++++--- sound/soc/sh/rcar/rsnd.h | 4 + sound/soc/soc-core.c | 267 +++++++++++++----- 11 files changed, 374 insertions(+), 164 deletions(-) -- 2.25.1

3 19

[PATCH 0/4] Add acpi mahine id's for vangogh platform and
by Venkata Prasad Potturu 12 Jul '23

12 Jul '23

This patch series is to add acpi machine id's for vangogh platform and add new dmi entries and machine driver support for nau8821 and max98388 codecs. Venkata Prasad Potturu (4): ASoC: amd: acp: Add machine driver support for nau8821 codec ASoC: amd: acp: Add machine driver support for max98388 codec ASoC: amd: Add acpi machine id's for vangogh platform ASoC: amd: Add new dmi entries to config entry sound/soc/amd/acp-config.c | 33 ++++ sound/soc/amd/acp/Kconfig | 2 + sound/soc/amd/acp/acp-mach-common.c | 267 ++++++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 2 + sound/soc/amd/acp/acp-sof-mach.c | 16 ++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 321 insertions(+) -- 2.25.1

2 5

[PATCH 0/2] ASoC: Another set of platform remove conversions
by Uwe Kleine-König 12 Jul '23

12 Jul '23

Hello, two more drivers were added during the current merge window that are users of the original .remove callback that I plan to get rid of. Convert them to .remove_new. Best regards Uwe Uwe Kleine-König (2): ASoC: amd: ps-sdw-dma: Convert to platform remove callback returning void ASoC: starfive: jh7110_tdm: Convert to platform remove callback returning void sound/soc/amd/ps/ps-sdw-dma.c | 5 ++--- sound/soc/starfive/jh7110_tdm.c | 5 ++--- 2 files changed, 4 insertions(+), 6 deletions(-) base-commit: 5133c9e51de41bfa902153888e11add3342ede18 -- 2.39.2

2 3

[PATCH 3/3] ASoC: rt5645: add the system level suspend-resume callback
by shumingf＠realtek.com 12 Jul '23

12 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> This patch handles the regmap settings and re-detects the jack when the system level suspend/resume. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5645.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index b0953e9bcaf9..5be5ec0260e9 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4199,11 +4199,44 @@ static void rt5645_i2c_shutdown(struct i2c_client *i2c) regmap_write(rt5645->regmap, RT5645_RESET, 0); } +static int __maybe_unused rt5645_sys_suspend(struct device *dev) +{ + struct rt5645_priv *rt5645 = dev_get_drvdata(dev); + + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); + cancel_delayed_work_sync(&rt5645->rcclock_work); + + regcache_cache_only(rt5645->regmap, true); + regcache_mark_dirty(rt5645->regmap); + return 0; +} + +static int __maybe_unused rt5645_sys_resume(struct device *dev) +{ + struct rt5645_priv *rt5645 = dev_get_drvdata(dev); + + regcache_cache_only(rt5645->regmap, false); + regcache_sync(rt5645->regmap); + + if (rt5645->hp_jack) { + rt5645->jack_type = 0; + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, msecs_to_jiffies(0)); + } + return 0; +} + +static const struct dev_pm_ops rt5645_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt5645_sys_suspend, rt5645_sys_resume) +}; + static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", .of_match_table = of_match_ptr(rt5645_of_match), .acpi_match_table = ACPI_PTR(rt5645_acpi_match), + .pm = &rt5645_pm, }, .probe = rt5645_i2c_probe, .remove = rt5645_i2c_remove, -- 2.41.0

2 1

[PATCH 1/3] ASoC: rt5645: implement set_jack callback
by shumingf＠realtek.com 12 Jul '23

12 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> Add a wrapper function to support set_jack component driver callback. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5645.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index acc7fb1581b2..c0d56a8f7f47 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3258,6 +3258,22 @@ int rt5645_set_jack_detect(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(rt5645_set_jack_detect); +static int rt5645_component_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct snd_soc_jack *mic_jack = NULL; + struct snd_soc_jack *btn_jack = NULL; + int *type = (int *)data; + + if (*type & SND_JACK_MICROPHONE) + mic_jack = hs_jack; + if (*type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + btn_jack = hs_jack; + + return rt5645_set_jack_detect(component, hs_jack, mic_jack, btn_jack); +} + static void rt5645_jack_detect_work(struct work_struct *work) { struct rt5645_priv *rt5645 = @@ -3532,6 +3548,7 @@ static const struct snd_soc_component_driver soc_component_dev_rt5645 = { .num_dapm_widgets = ARRAY_SIZE(rt5645_dapm_widgets), .dapm_routes = rt5645_dapm_routes, .num_dapm_routes = ARRAY_SIZE(rt5645_dapm_routes), + .set_jack = rt5645_component_set_jack, .use_pmdown_time = 1, .endianness = 1, }; -- 2.41.0

2 1

[PATCH v3 0/3] AMD Vangogh support for NAU8821/MAX98388
by Cristian Ciocaltea 12 Jul '23

12 Jul '23

This patch series extends the Vangogh machine driver to support a variant based on the Nuvoton NAU88L21 Codec and the Analog Devices MAX98388 Speaker Amplifier. Changes in v3: * Dropped acp5x_max98388_hw_params() in PATCH 3/3 and rely on dai_link to set format, according to Mark's review * v2: https://lore.kernel.org/all/20230705134341.175889-1-cristian.ciocaltea@coll… Changes in v2: * Reworked series to ensure the existing devices based on the CS35L41 codec continue to work fine * Rebased onto next-20230703 Cristian Ciocaltea (3): ASoC: amd: vangogh: Make use of DRV_NAME ASoC: amd: vangogh: Use dmi_first_match() for DMI quirk handling ASoC: amd: vangogh: Add support for NAU8821/MAX98388 variant sound/soc/amd/Kconfig | 5 +- sound/soc/amd/vangogh/acp5x-mach.c | 149 +++++++++++++++++++++++++---- 2 files changed, 131 insertions(+), 23 deletions(-) -- 2.41.0

2 4

[PATCH v2 0/4] ASoC: remove copy of intlog10()
by Andy Shevchenko 12 Jul '23

12 Jul '23

The first three patches moves intlog10() to be available in entire kernel. The last one removes copy of it in one driver. Besides already good Lines of Code (LoC) statistics the upcoming users, if any, can utilize the exported functions. The series can be routed via ASoC tree (as Mauro suggested). Note, int_log.h is separated from math.h due to licensing. I dunno if we can mix two in a single header file. In any case we may do it later on. Changelog v2: - moved kernel doc inclusion to the generic place (Mauro) - added tags (Mauro, Mark) Andy Shevchenko (4): lib/math: Move dvb_math.c into lib/math/int_log.c lib/math/int_log: Use ARRAY_SIZE(logtable) where makes sense lib/math/int_log: Replace LGPL-2.1-or-later boilerplate with SPDX identifier ASoC: nau8825: Replace copied'n'pasted intlog10() Documentation/core-api/kernel-api.rst | 7 +- Documentation/driver-api/media/dtv-common.rst | 9 -- drivers/media/dvb-core/Makefile | 2 +- drivers/media/dvb-frontends/af9013_priv.h | 2 +- drivers/media/dvb-frontends/af9033_priv.h | 2 +- drivers/media/dvb-frontends/cxd2820r_priv.h | 2 +- drivers/media/dvb-frontends/cxd2841er.c | 2 +- .../cxd2880/cxd2880_tnrdmd_dvbt2_mon.c | 2 +- .../cxd2880/cxd2880_tnrdmd_dvbt_mon.c | 2 +- .../media/dvb-frontends/cxd2880/cxd2880_top.c | 2 +- drivers/media/dvb-frontends/dib7000p.c | 2 +- drivers/media/dvb-frontends/dib8000.c | 2 +- drivers/media/dvb-frontends/dib9000.c | 2 +- drivers/media/dvb-frontends/drxk_hard.c | 2 +- drivers/media/dvb-frontends/lgdt3305.c | 2 +- drivers/media/dvb-frontends/lgdt3306a.c | 2 +- drivers/media/dvb-frontends/lgdt330x.c | 2 +- drivers/media/dvb-frontends/m88ds3103_priv.h | 2 +- drivers/media/dvb-frontends/mn88443x.c | 2 +- drivers/media/dvb-frontends/mn88472_priv.h | 2 +- drivers/media/dvb-frontends/mn88473_priv.h | 2 +- drivers/media/dvb-frontends/or51132.c | 2 +- drivers/media/dvb-frontends/or51211.c | 2 +- drivers/media/dvb-frontends/rtl2830_priv.h | 2 +- drivers/media/dvb-frontends/rtl2832_priv.h | 2 +- drivers/media/dvb-frontends/si2165.c | 2 +- drivers/media/dvb-frontends/stv0367.c | 2 +- drivers/media/dvb-frontends/tc90522.c | 2 +- drivers/media/dvb-frontends/tda10048.c | 2 +- include/{media/dvb_math.h => linux/int_log.h} | 18 +--- lib/math/Makefile | 2 +- .../dvb-core/dvb_math.c => lib/math/int_log.c | 26 ++---- sound/soc/codecs/nau8825.c | 93 +------------------ 33 files changed, 49 insertions(+), 160 deletions(-) rename include/{media/dvb_math.h => linux/int_log.h} (63%) rename drivers/media/dvb-core/dvb_math.c => lib/math/int_log.c (84%) -- 2.40.0.1.gaa8946217a0b

3 6

[PATCH 1/2] ASoC: SOF: amd: add revision check for sending sha dma completion command
by Mastan Katragadda 12 Jul '23

12 Jul '23

ACP driver should send SHA DMA completion command to PSP module for RN platform only. Add a revision check for RN platform. Signed-off-by: Mastan Katragadda <Mastan.Katragadda(a)amd.com> --- sound/soc/sof/amd/acp.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 2ae76bcd3590..afb505461ea1 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -217,6 +217,7 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, unsigned int image_length) { struct snd_sof_dev *sdev = adata->dev; + const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int tx_count, fw_qualifier, val; int ret; @@ -251,9 +252,12 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - ret = psp_send_cmd(adata, MBOX_ACP_SHA_DMA_COMMAND); - if (ret) - return ret; + /* psp_send_cmd only required for renoir platform (rev - 3) */ + if (desc->rev == 3) { + ret = psp_send_cmd(adata, MBOX_ACP_SHA_DMA_COMMAND); + if (ret) + return ret; + } ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_DSP_FW_QUALIFIER, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, -- 2.25.1

3 9

[PATCH v2] ASoC: stac9766: fix build errors with REGMAP_AC97
by Randy Dunlap 12 Jul '23

12 Jul '23

Select REGMAP_AC97 to fix these build errors: ERROR: modpost: "regmap_ac97_default_volatile" [sound/soc/codecs/snd-soc-stac9766.ko] undefined! ERROR: modpost: "__regmap_init_ac97" [sound/soc/codecs/snd-soc-stac9766.ko] undefined! Fixes: 6bbf787bb70c ("ASoC: stac9766: Convert to regmap") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org --- v2: rebase/resend sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff -- a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1707,6 +1707,7 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate depends on SND_SOC_AC97_BUS + select REGMAP_AC97 config SND_SOC_STI_SAS tristate "codec Audio support for STI SAS codec"

2 1

[PATCH 0/2] ASoC: mt8188: add memory-region support
by Trevor Wu 12 Jul '23

12 Jul '23

This series introduces support for memory-region, allowing afe memif to utilize the region specified in dts. Trevor Wu (2): ASoC: mediatek: mt8188: add memory-region support ASoC: dt-bindings: mediatek,mt8188-afe: add memory-region .../devicetree/bindings/sound/mediatek,mt8188-afe.yaml | 7 +++++++ sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 8 ++++++-- 2 files changed, 13 insertions(+), 2 deletions(-) -- 2.18.0

3 4

[PATCH] ASoC: soc-core: protect dlc->of_node under mutex
by Kuninori Morimoto 12 Jul '23

12 Jul '23

dlc->of_node will be set on snd_soc_get_dlc(), but we want 1) protect it by mutex, 2) set only when successed. This patch do it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 11bc5250ffd0..fcc798e0b8f0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3262,8 +3262,6 @@ int snd_soc_get_dlc(const struct of_phandle_args *args, struct snd_soc_dai_link_ struct snd_soc_component *pos; int ret = -EPROBE_DEFER; - dlc->of_node = args->np; - mutex_lock(&client_mutex); for_each_component(pos) { struct device_node *component_of_node = soc_component_to_node(pos); @@ -3317,6 +3315,10 @@ int snd_soc_get_dlc(const struct of_phandle_args *args, struct snd_soc_dai_link_ break; } + + if (ret == 0) + dlc->of_node = args->np; + mutex_unlock(&client_mutex); return ret; } -- 2.25.1

2 1

[PATCH 0/8] ASoC: Intel: avs: New boards and fixes to existing ones
by Cezary Rojewski 12 Jul '23

12 Jul '23

Series adds support for two boards: es8336 and rt5663. The former is utilized by some KBL-based tablets whereas the latter unlocks Chromebooks with rt5663 i2c codecs. As existing implementation of es8336 (es8316.c) codec driver is not prepared to cope with KBL-based platforms, couple of small, clock-related changes precede anything avs-driver related. The tail of patchset cleans up existing implementation of rt5682. Amadeusz Sławiński (2): ASoC: Intel: avs: Add rt5663 machine board ASoC: Intel: avs: Load rt5663 board on KBL-based platforms Cezary Rojewski (6): ASoC: codecs: es8316: Add support for 24 MHz MCLK ASoC: codecs: es8316: Add support for S24_3LE format ASoC: Intel: avs: Add es8336 machine board ASoC: Intel: avs: Load es8336 board on KBL-based platforms ASoC: Intel: avs: rt5682: Add missing components ASoC: Intel: avs: rt5682: Tidy up hw_params() sound/soc/codecs/es8316.c | 5 +- sound/soc/intel/avs/board_selection.c | 16 ++ sound/soc/intel/avs/boards/Kconfig | 20 ++ sound/soc/intel/avs/boards/Makefile | 4 + sound/soc/intel/avs/boards/es8336.c | 315 ++++++++++++++++++++++++++ sound/soc/intel/avs/boards/rt5663.c | 254 +++++++++++++++++++++ sound/soc/intel/avs/boards/rt5682.c | 67 +++--- 7 files changed, 652 insertions(+), 29 deletions(-) create mode 100644 sound/soc/intel/avs/boards/es8336.c create mode 100644 sound/soc/intel/avs/boards/rt5663.c -- 2.25.1

2 9

[PATCH] ASoC: imx-pcm-rpmsg: Set PCM hardware parameters separately
by Chancel Liu 12 Jul '23

12 Jul '23

Different PCM devices may have different PCM hardware parameters. It requires PCM hardware parameters set separately if there is more than one rpmsg sound card. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/imx-pcm-rpmsg.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index 765dad607bf6..d63782b8bdef 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -228,6 +228,10 @@ static int imx_rpmsg_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct rpmsg_info *info = dev_get_drvdata(component->dev); + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev); + struct snd_pcm_hardware pcm_hardware; struct rpmsg_msg *msg; int ret = 0; int cmd; @@ -255,10 +259,11 @@ static int imx_rpmsg_pcm_open(struct snd_soc_component *component, info->send_message(msg, info); - imx_rpmsg_pcm_hardware.period_bytes_max = - imx_rpmsg_pcm_hardware.buffer_bytes_max / 2; + pcm_hardware = imx_rpmsg_pcm_hardware; + pcm_hardware.buffer_bytes_max = rpmsg->buffer_size; + pcm_hardware.period_bytes_max = pcm_hardware.buffer_bytes_max / 2; - snd_soc_set_runtime_hwparams(substream, &imx_rpmsg_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, &pcm_hardware); ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -597,7 +602,6 @@ static int imx_rpmsg_pcm_new(struct snd_soc_component *component, if (ret) return ret; - imx_rpmsg_pcm_hardware.buffer_bytes_max = rpmsg->buffer_size; return snd_pcm_set_fixed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV_WC, pcm->card->dev, rpmsg->buffer_size); } -- 2.25.1

3 2

[PATCH] ASoC: dt-bindings: nau8821: Convert to dtschema
by Seven Lee 12 Jul '23

12 Jul '23

Convert the NAU8821 audio CODEC bindings to DT schema. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 55 -------- .../bindings/sound/nuvoton,nau8821.yaml | 125 ++++++++++++++++++ 2 files changed, 125 insertions(+), 55 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt deleted file mode 100644 index 7c84e7c7327a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8821.txt +++ /dev/null @@ -1,55 +0,0 @@ -Nuvoton NAU88L21 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8821" - - - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. - - nuvoton,key_enable: Headset button detection switch. - -Example: - - headset: nau8821@1b { - compatible = "nuvoton,nau8821"; - reg = <0x1b>; - interrupt-parent = <&gpio>; - interrupts = <23 IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,dmic-clk-threshold = 3072000; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml new file mode 100644 index 000000000000..fc2f4ce4db88 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -0,0 +1,125 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8821.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU88L21 audio codec + +maintainers: + - Seven Lee <wtli(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8821 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: Pull-up JKDET pin. If set then JKDET pin is pull up, + otherwise pull down. + type: boolean + + nuvoton,key-enable: + description: handles key press detection. + type: boolean + + nuvoton,jkdet-polarity: + description: JKDET pin polarity. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,micbias-voltage: + description: MICBIAS output level select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # VDDA + - 1 # VDDA * 1 + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,vref-impedance: + description: VMID Tie-off impedance select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # open + - 1 # 25KOhms + - 2 # 125KOhms + - 3 # 2.5KOhms + default: 2 + + nuvoton,jack-insert-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,dmic-clk-threshold: + description: DMIC clock speed expected value. Unit is Hz. + $ref: /schemas/types.yaml#/definitions/uint32 + default: 3072000 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1b { + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,key-enable; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,micbias-voltage = <6>; + nuvoton,vref-impedance = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <0>; + nuvoton,dmic-clk-threshold = <3072000>; + #sound-dai-cells = <0>; + }; + }; -- 2.25.1

4 4

[PATCH v6 00/13] Add support for IIO devices in ASoC
by Herve Codina 12 Jul '23

12 Jul '23

Several weeks ago, I sent a series [1] for adding a potentiometer as an auxiliary device in ASoC. The feedback was that the potentiometer should be directly handled in IIO (as other potentiometers) and something more generic should be present in ASoC in order to have a binding to import some IIO devices into sound cards. The series related to the IIO potentiometer device is already applied. This series introduces audio-iio-aux. Its goal is to offer the binding between IIO and ASoC. It exposes attached IIO devices as ASoC auxiliary devices and allows to control them through mixer controls. On my system, the IIO device is a potentiometer and it is present in an amplifier design present in the audio path. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230615152631.224529-1-herve.codina@b… This v6 series mainly: - Fixes comments - Does some minor code modification Best regards, Hervé [1] https://lore.kernel.org/linux-kernel/20230203111422.142479-1-herve.codina@b… [2] https://lore.kernel.org/linux-kernel/20230421085245.302169-1-herve.codina@b… Changes v5 -> v6 - Patches 3, 4, 11, 13 No changes - Patch 1 Add 'Reviewed-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>' Add 'Reviewed-by: Rob Herring <robh(a)kernel.org>' - Patch 2 Add 'Reviewed-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>' - Patches 5, 6 Add 'Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>' - Patch 7 Fix comments: 'buf' -> 'buff', '__unconst_typeof' -> '__unconst_integer_typeof' - Patch 8, 9, 10 Add 'Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>' - Patch 12 Set struct snd_kcontrol_new fields for the constrol variable at variable initalization. Add a missing blank line Add a comment related to min/max swapping Add 'Reviewed-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>' Changes v4 -> v5 - Patches 1, 2, 3, 4, 5, 9, 10, 11, 12, 13 No changes. - Patch 6 Fix commit log. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' - Patch 7 Fix the macros to be able to use them with: - an array defined as int *buff; - an array defined as int buff[N]; - Rework the way to "unconstify" the temporary variable to avoid issues due to integer promotion. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' - Patch 8 Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' Changes v3 -> v4 - Patches 1, 2 No changes. - Patches 3, 4, 5 Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 6 (new in v4) Fix headers inclusion order. - Patch 7 (patch 6 in v3) Add a comment related to __must_be_array() Use __array[0] of *__array - Patch 8 (patch 7 in v3) Fix minmax.h inclusion order. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 9 (patch 8 in v3) Add 'Suggested-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 10 (patch 9 in v3) Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 11 (patch 10 in v3) Fix a typo. Add 'Suggested-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 12 (patch 11 in v3) Fix typos in the commit log. Fix headers inclusion order. Removed unneeded variable initialization. Replace {0} by {}. Use struct device *dev in probe(). Check an error on the snd-control-invert-range property read. - Patch 13 (patch12 in v3) No changes. Changes v2 -> v3 - Patches 1, 2 No changes. - Patch 3, 4 Add 'Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. - Patch 5 (new in v3) Removed the 'unused' variable and check the null pointer when used. - Patch 6 (new in v3) Introduce {min,max}_array(). - Patch 7 (new in v3) Use max_array() in iio_channel_read_max(). - Patch 8 (new in v3) Replace a FIXME comment by a TODO one. - Patch 9 (patch 5 in v2) Removed the 'unused' variable and check the null pointer when used. Use min_array(). Remplace a FIXME comment by a TODO one. - Patch 10 (patch 6 in v2) Convert existing macros to return a compound litteral instead of adding a new helper. - Patch 11 (patch 7 in v2) Remove the file name from the C file header. Use directly converted DAPM macros. Replace <linux/module.h> by <linux/mod_devicetable.h>. Add <linux/platform_device.h>. Be sure that min <= max. Swap values if it is not the case. Move the bool structure member after the int ones. Remove unneeded assignements. Use dev_err_probe() when relevant. Use str_on_off(). Use static_assert() instead of BUILD_BUG_ON(). Remove unneeded comma and blank line. Use device_property_*() instead of the OF API. - patch 8 available in v2 removed as already applied - Patch 12 (patch 9 in v2) Use devm_add_action_or_reset(). Call simple_populate_aux() from simple_parse_of(). Changes v1 -> v2 - Patch 1 Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the /schemas/iio/iio-consumer.yaml reference Remove the unneeded '|' after description - Patch 2 (new in v2) Introduce the simple-audio-card additional-devs subnode - Patch 3 (new in v2) Check err before switch() in iio_channel_read_max() - Patch 4 (new in v2) Fix raw reads and raw writes documentation - Patch 5 (patch 2 in v1) Check err before switch() in iio_channel_read_min() Fix documentation - Patch 6 (path 3 in v1) No changes - Patch 7 (patch 4 in v1) Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the mask usage from audio_iio_aux_{get,put}_volsw helpers Use directly PTR_ERR() in dev_err_probe() parameter Remove the '!!' construction Remove of_match_ptr() - Patch 8 (new in v2) Add a missing of_node_put() in the simple-card driver - Patch 9 (new in v2) Handle additional-devs in the simple-card driver Herve Codina (13): ASoC: dt-bindings: Add audio-iio-aux ASoC: dt-bindings: simple-card: Add additional-devs subnode iio: inkern: Check error explicitly in iio_channel_read_max() iio: consumer.h: Fix raw values documentation notes iio: inkern: Remove the 'unused' variable usage in iio_channel_read_max() iio: inkern: Fix headers inclusion order minmax: Introduce {min,max}_array() iio: inkern: Use max_array() to get the maximum value from an array iio: inkern: Replace a FIXME comment by a TODO one iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Convert macros to return a compound literal ASoC: codecs: Add support for the generic IIO auxiliary devices ASoC: simple-card: Handle additional devices .../bindings/sound/audio-iio-aux.yaml | 64 ++++ .../bindings/sound/simple-card.yaml | 53 +++ drivers/iio/inkern.c | 86 ++++- include/linux/iio/consumer.h | 37 +- include/linux/minmax.h | 64 ++++ include/sound/soc-dapm.h | 138 ++++--- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/audio-iio-aux.c | 344 ++++++++++++++++++ sound/soc/generic/simple-card.c | 46 ++- 10 files changed, 775 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-iio-aux.yaml create mode 100644 sound/soc/codecs/audio-iio-aux.c -- 2.40.1

3 29

[PATCH v2 00/10] Refactor acp legacy driver and add
by Syed Saba Kareem 12 Jul '23

12 Jul '23

This patch series to refactor acp leagacy driver and add pm ops support for rembrandt platforms. Changes since V1 - Fix below kernel test robot errors and warnings - ld.lld: error: undefined symbol: acp_enable_interrupts - referenced by acp-renoir.c:191 (sound/soc/amd/acp/acp-renoir.c:191) - ld.lld: error: undefined symbol: acp_disable_interrupts - sound/soc/amd/acp/acp-renoir.o:(renoir_audio_remove) in archive vmlinux.a - All warnings (new ones prefixed by >>): - sound/soc/amd/acp/acp-rembrandt.c:169:5: warning: no previous - prototype for 'acp6x_master_clock_generate' [-Wmissing-prototypes] - 169 | int acp6x_master_clock_generate(struct device *dev) Syed Saba Kareem (10): ASoC: amd: acp: refactor the acp init and de-init sequence ASoC: amd: acp: add acp i2s master clock generation for rembrandt platform ASoC: amd: acp: remove the redundant acp enable/disable interrupts functions ASoC: amd: acp: store platform device reference created in pci probe call ASoC: amd: acp: add pm ops support for acp pci driver ASoC: amd: acp: store xfer_resolution of the stream ASoC: amd: acp: export config_acp_dma() and config_pte_for_stream() symbols ASoC: amd: acp: store the pdm stream channel mask ASoC: amd: acp: move pdm macros to common header file ASoC: amd: acp: add pm ops support for rembrandt platform sound/soc/amd/acp/Kconfig | 5 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-i2s.c | 2 + sound/soc/amd/acp/acp-legacy-common.c | 347 ++++++++++++++++++++++++++ sound/soc/amd/acp/acp-pci.c | 57 ++++- sound/soc/amd/acp/acp-pdm.c | 13 +- sound/soc/amd/acp/acp-platform.c | 6 +- sound/soc/amd/acp/acp-rembrandt.c | 186 +++++--------- sound/soc/amd/acp/acp-renoir.c | 115 +-------- sound/soc/amd/acp/amd.h | 51 ++++ 10 files changed, 531 insertions(+), 253 deletions(-) create mode 100644 sound/soc/amd/acp/acp-legacy-common.c -- 2.25.1

4 13

[PATCH] ASoC: rt722-sdca: Remove redundant sdca mask
by Jack Yu 12 Jul '23

12 Jul '23

Remove redundant sdca mask for clear code. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca.c | 3 +-- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index cc57e4e27805..a9416bd163e8 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -175,7 +175,7 @@ static int rt722_sdca_update_status(struct sdw_slave *slave, * This also could sync with the cache value as the rt722_sdca_jack_init set. */ sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_6); sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); } diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index 9c0d34366c9e..0e1c65a20392 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -191,8 +191,7 @@ static void rt722_sdca_jack_detect_handler(struct work_struct *work) return; /* SDW_SCP_SDCA_INT_SDCA_6 is used for jack detection */ - if (rt722->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6 || - rt722->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_0) { + if (rt722->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { ret = rt722_sdca_headset_detect(rt722); if (ret < 0) return; -- 2.40.1

2 3

[PATCH 1/1] ASoC: dwc: Add TDM mode support
by Maxim Kochetkov 12 Jul '23

12 Jul '23

Depending on hardware implementaion of DWC I2S controller may support TDM mode if enabled in SoC at design time. Unfortunately there is no way to detect TDM capability for DWC by reading registers. Anyway, if such capability enabled, TDM mode can be enabled and configured by dai-tdm-slot-* DT options. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/dwc/dwc-i2s.c | 65 +++++++++++++++++++++++++++++++++++++++-- sound/soc/dwc/dwc-pcm.c | 8 ++--- sound/soc/dwc/local.h | 24 +++++++++++++++ 3 files changed, 90 insertions(+), 7 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 97d652f0e84d..1f1ee14b04e6 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -183,7 +183,15 @@ static void i2s_start(struct dw_i2s_dev *dev, { struct i2s_clk_config_data *config = &dev->config; - i2s_write_reg(dev->i2s_base, IER, 1); + u32 reg = IER_IEN; + + if (dev->tdm_slots) { + reg |= (dev->tdm_slots - 1) << IER_TDM_SLOTS_SHIFT; + reg |= IER_INTF_TYPE; + reg |= dev->frame_offset << IER_FRAME_OFF_SHIFT; + } + + i2s_write_reg(dev->i2s_base, IER, reg); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) i2s_write_reg(dev->i2s_base, ITER, 1); @@ -233,13 +241,15 @@ static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) dev->xfer_resolution); i2s_write_reg(dev->i2s_base, TFCR(ch_reg), dev->fifo_th - 1); - i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + i2s_write_reg(dev->i2s_base, TER(ch_reg), TER_TXCHEN | + dev->tdm_mask << TER_TXSLOT_SHIFT); } else { i2s_write_reg(dev->i2s_base, RCR(ch_reg), dev->xfer_resolution); i2s_write_reg(dev->i2s_base, RFCR(ch_reg), dev->fifo_th - 1); - i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + i2s_write_reg(dev->i2s_base, RER(ch_reg), RER_RXCHEN | + dev->tdm_mask << RER_RXSLOT_SHIFT); } } @@ -276,6 +286,9 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + if (dev->tdm_slots) + config->data_width = 32; + config->chan_nr = params_channels(params); switch (config->chan_nr) { @@ -384,14 +397,58 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ret = -EINVAL; break; } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_DSP_A: + dev->frame_offset = 1; + break; + case SND_SOC_DAIFMT_DSP_B: + dev->frame_offset = 0; + break; + default: + dev_err(dev->dev, "DAI format unsupported"); + return -EINVAL; + } + return ret; } +static int dw_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + + if (slot_width != 32) + return -EINVAL; + + if (slots < 0 || slots > 16) + return -EINVAL; + + if (rx_mask != tx_mask) + return -EINVAL; + + if (!rx_mask) + return -EINVAL; + + dev->tdm_slots = slots; + dev->tdm_mask = rx_mask; + + dev->l_reg = RSLOT_TSLOT(ffs(rx_mask) - 1); + dev->r_reg = RSLOT_TSLOT(fls(rx_mask) - 1); + + return 0; +} + static const struct snd_soc_dai_ops dw_i2s_dai_ops = { .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, .set_fmt = dw_i2s_set_fmt, + .set_tdm_slot = dw_i2s_set_tdm_slot, }; #ifdef CONFIG_PM @@ -726,6 +783,8 @@ static int dw_i2s_probe(struct platform_device *pdev) if (irq >= 0) { ret = dw_pcm_register(pdev); dev->use_pio = true; + dev->l_reg = LRBR_LTHR(0); + dev->r_reg = RRBR_RTHR(0); } else { ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); diff --git a/sound/soc/dwc/dwc-pcm.c b/sound/soc/dwc/dwc-pcm.c index 9f25631d43d3..f99262b89008 100644 --- a/sound/soc/dwc/dwc-pcm.c +++ b/sound/soc/dwc/dwc-pcm.c @@ -31,8 +31,8 @@ static unsigned int dw_pcm_tx_##sample_bits(struct dw_i2s_dev *dev, \ int i; \ \ for (i = 0; i < dev->fifo_th; i++) { \ - iowrite32(p[tx_ptr][0], dev->i2s_base + LRBR_LTHR(0)); \ - iowrite32(p[tx_ptr][1], dev->i2s_base + RRBR_RTHR(0)); \ + iowrite32(p[tx_ptr][0], dev->i2s_base + dev->l_reg); \ + iowrite32(p[tx_ptr][1], dev->i2s_base + dev->r_reg); \ period_pos++; \ if (++tx_ptr >= runtime->buffer_size) \ tx_ptr = 0; \ @@ -51,8 +51,8 @@ static unsigned int dw_pcm_rx_##sample_bits(struct dw_i2s_dev *dev, \ int i; \ \ for (i = 0; i < dev->fifo_th; i++) { \ - p[rx_ptr][0] = ioread32(dev->i2s_base + LRBR_LTHR(0)); \ - p[rx_ptr][1] = ioread32(dev->i2s_base + RRBR_RTHR(0)); \ + p[rx_ptr][0] = ioread32(dev->i2s_base + dev->l_reg); \ + p[rx_ptr][1] = ioread32(dev->i2s_base + dev->r_reg); \ period_pos++; \ if (++rx_ptr >= runtime->buffer_size) \ rx_ptr = 0; \ diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index ba4e397099be..4ce96bac2f39 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -25,6 +25,13 @@ #define RXFFR 0x014 #define TXFFR 0x018 +/* Enable register fields */ +#define IER_TDM_SLOTS_SHIFT 8 +#define IER_FRAME_OFF_SHIFT 5 +#define IER_FRAME_OFF BIT(5) +#define IER_INTF_TYPE BIT(1) +#define IER_IEN BIT(0) + /* Interrupt status register fields */ #define ISR_TXFO BIT(5) #define ISR_TXFE BIT(4) @@ -46,6 +53,15 @@ #define TFCR(x) (0x40 * x + 0x04C) #define RFF(x) (0x40 * x + 0x050) #define TFF(x) (0x40 * x + 0x054) +#define RSLOT_TSLOT(x) (0x4 * (x) + 0x224) + +/* Receive enable register fields */ +#define RER_RXSLOT_SHIFT 8 +#define RER_RXCHEN BIT(0) + +/* Transmit enable register fields */ +#define TER_TXSLOT_SHIFT 8 +#define TER_TXCHEN BIT(0) /* I2SCOMPRegisters */ #define I2S_COMP_PARAM_2 0x01F0 @@ -105,6 +121,8 @@ struct dw_i2s_dev { u32 ccr; u32 xfer_resolution; u32 fifo_th; + u32 l_reg; + u32 r_reg; /* data related to DMA transfers b/w i2s and DMAC */ union dw_i2s_snd_dma_data play_dma_data; @@ -114,6 +132,12 @@ struct dw_i2s_dev { /* data related to PIO transfers */ bool use_pio; + + /* data related to TDM mode */ + u32 tdm_slots; + u32 tdm_mask; + u32 frame_offset; + struct snd_pcm_substream __rcu *tx_substream; struct snd_pcm_substream __rcu *rx_substream; unsigned int (*tx_fn)(struct dw_i2s_dev *dev, -- 2.40.1

2 3

[PATCH -next] ASoC: tas2781: No need to set device_driver owner
by Yang Li 12 Jul '23

12 Jul '23

Remove .owner field if calls are used which set it automatically. to silence the warning: ./sound/soc/codecs/tas2781-i2c.c:746:3-8: No need to set .owner here. The core will do it. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Closes: https://bugzilla.openanolis.cn/show_bug.cgi?id=5589 Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/codecs/tas2781-i2c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 4c59429a42b7..55cd5e3c23a5 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -743,7 +743,6 @@ MODULE_DEVICE_TABLE(acpi, tasdevice_acpi_match); static struct i2c_driver tasdevice_i2c_driver = { .driver = { .name = "tas2781-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tasdevice_of_match), #ifdef CONFIG_ACPI .acpi_match_table = ACPI_PTR(tasdevice_acpi_match), -- 2.20.1.7.g153144c

2 1

[PATCH v3] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 12 Jul '23

12 Jul '23

This configuration supports JSL boards which implement ALC5650 dual I2S interface codec. Two DAI links are added: AIF1 (on codec side) for headphone and AIF2 for speakers. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/sof_rt5682.c | 80 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 3 files changed, 93 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f472f603ab75..1fe830af2b84 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -475,7 +475,7 @@ endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC if SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_SOF_RT5682_MACH - tristate "SOF with rt5682 codec in I2S Mode" + tristate "SOF with rt5650/rt5682 codec in I2S Mode" depends on I2C && ACPI depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ @@ -485,6 +485,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P + select SND_SOC_RT5645 select SND_SOC_RT5682_I2C select SND_SOC_RT5682S select SND_SOC_DMIC @@ -494,7 +495,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_INTEL_SOF_REALTEK_COMMON help This adds support for ASoC machine driver for SOF platforms - with rt5682 codec. + with rt5650 or rt5682 codec. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7c034d671cf3..b4f07bdcf8b4 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -22,6 +22,7 @@ #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" +#include "../../codecs/rt5645.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" @@ -60,6 +61,7 @@ #define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) #define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) #define SOF_RT1019_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_RT5650_HEADPHONE_CODEC_PRESENT BIT(27) /* Default: MCLK on, MCLK 19.2M, SSP0 */ @@ -305,6 +307,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; + int extra_jack_data; int ret; /* need to enable ASRC function for 24MHz mclk rate */ @@ -315,7 +318,16 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) RT5682S_DA_STEREO1_FILTER | RT5682S_AD_STEREO1_FILTER, RT5682S_CLK_SEL_I2S1_ASRC); - else + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(component, + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | RT5682_AD_STEREO1_FILTER, @@ -365,7 +377,12 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); + + if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + extra_jack_data = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra_jack_data); + } else + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); @@ -402,6 +419,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_MCLK; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_MCLK; else pll_source = RT5682_PLL1_S_MCLK; @@ -422,6 +441,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, } else { if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_BCLK1; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_BCLK1; else pll_source = RT5682_PLL1_S_BCLK1; @@ -431,6 +452,9 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + pll_id = 0; /* not used in codec driver */ + clk_id = RT5645_SCLK_S_PLL1; } else { pll_id = RT5682_PLL1; clk_id = RT5682_SCLK_S_PLL1; @@ -559,11 +583,30 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "SPOL" }, + { "Right Spk", NULL, "SPOR" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes, + ARRAY_SIZE(rt5650_spk_dapm_routes)); + if (ret) + dev_err(rtd->dev, "fail to add dapm routes, ret=%d\n", ret); + + return ret; +} + static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -614,6 +657,17 @@ static struct snd_soc_dai_link_component rt5682s_component[] = { } }; +static struct snd_soc_dai_link_component rt5650_components[] = { + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif1", + }, + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif2", + } +}; + static struct snd_soc_dai_link_component dmic_component[] = { { .name = "dmic-codec", @@ -652,6 +706,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { links[id].codecs = rt5682s_component; links[id].num_codecs = ARRAY_SIZE(rt5682s_component); + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[0]; + links[id].num_codecs = 1; } else { links[id].codecs = rt5682_component; links[id].num_codecs = ARRAY_SIZE(rt5682_component); @@ -804,6 +861,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].init = max_98390_spk_codec_init; links[id].ops = &max_98390_ops; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[1]; + links[id].num_codecs = 1; + links[id].init = rt5650_spk_init; + links[id].ops = &sof_rt5682_ops; } else { max_98357a_dai_link(&links[id]); } @@ -890,6 +952,12 @@ static int sof_audio_probe(struct platform_device *pdev) /* Detect the headset codec variant */ if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + else if (acpi_dev_present("10EC5650", NULL, -1)) { + sof_rt5682_quirk |= SOF_RT5650_HEADPHONE_CODEC_PRESENT; + + sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", + GFP_KERNEL); + } if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; @@ -1178,6 +1246,14 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(0) | SOF_RT5682_NUM_HDMIDEV(3)), }, + { + .name = "jsl_rt5650", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index f5c7e1bbded0..36ea2bacbe7d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -34,6 +34,11 @@ static const struct snd_soc_acpi_codecs mx98360a_spk = { .codecs = {"MX98360A"} }; +static struct snd_soc_acpi_codecs rt5650_spk = { + .num_codecs = 1, + .codecs = {"10EC5650"} +}; + static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {"10EC5682", "RTL5682"}, @@ -98,6 +103,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .id = "10EC5650", + .drv_name = "jsl_rt5650", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt5650_spk, + .sof_tplg_filename = "sof-jsl-rt5650-dual.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

3 4

[PATCH v2] ASoC: dt-bindings: Update maintainer email id
by Rohit kumar 12 Jul '23

12 Jul '23

Updated my mail id to latest quicinc id. Acked-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Signed-off-by: Rohit kumar <quic_rohkumar(a)quicinc.com> --- v2: Updated commit header .../devicetree/bindings/sound/google,sc7180-trogdor.yaml | 2 +- Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 666a95ac22c8..ba5b7728cf33 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Google SC7180-Trogdor ASoC sound card driver maintainers: - - Rohit kumar <rohitkr(a)codeaurora.org> + - Rohit kumar <quic_rohkumar(a)quicinc.com> - Cheng-Yi Chiang <cychiang(a)chromium.org> description: diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml index 6cc8f86c7531..3a559bd07a79 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -8,7 +8,7 @@ title: Qualcomm Technologies Inc. LPASS CPU dai driver maintainers: - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> - - Rohit kumar <rohitkr(a)codeaurora.org> + - Rohit kumar <quic_rohkumar(a)quicinc.com> description: | Qualcomm Technologies Inc. SOC Low-Power Audio SubSystem (LPASS) that consist -- 2.25.1

3 3

[PATCH v1] ASoC: dt-bindings: Add tas2505 audio amplifier
by Kevin Lu 12 Jul '23

12 Jul '23

Create tas2505.yaml for tas2505. Signed-off-by: Kevin Lu <luminlong(a)139.com> --- .../devicetree/bindings/sound/tas2505.yaml | 72 +++++++++++++++++++ 1 file changed, 72 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2505.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2505.yaml b/Documentation/devicetree/bindings/sound/tas2505.yaml new file mode 100644 index 0000000..6cc3fe4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2505.yaml @@ -0,0 +1,72 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2505.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2505 Mono Audio Amplifier + +maintainers: + - Kevin Lu <kevin-lu(a)ti.com> + +description: | + The TAS2505 is a mono Class-D speaker amp that supports both Digital + and Analog inputs. The device is ideal for automotive instrument cluster, + emergency call (eCall), and telematics applications. Direct I2S input + removes the need for an external DAC in the audio signal path. + An on-chip PLL provides the high speed clock needed by the DSP. + The volume level is register controlled. + + Specifications about the audio amplifier can be found at: + https://www.ti.com/lit/gpn/tas2505 + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,tas2505 + + reg: + maxItems: 1 + description: | + I2C address of the device can be set to 0x18. + + reset-gpios: + maxItems: 1 + description: GPIO used to reset the device. + + clock-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM clock sourse. + enum: + - 0 # Select MLCK + - 1 # Select BLCK + + '#sound-dai-cells': + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@18 { + compatible = "ti,tas2505"; + reg = <0x18>; + #sound-dai-cells = <0>; + reset-gpios = <&gpio4 21 GPIO_ACTIVE_LOW>; + clock-source = <0>; + }; + }; -- 2.34.1

3 2

[PATCH] ASoC: dt-bindings: nau8821: Convert to dtschema
by Seven Lee 12 Jul '23

12 Jul '23

Convert the NAU8821 audio CODEC bindings to DT schema. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 55 -------- .../bindings/sound/nuvoton,nau8821.yaml | 120 ++++++++++++++++++ 2 files changed, 120 insertions(+), 55 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt deleted file mode 100644 index 7c84e7c7327a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8821.txt +++ /dev/null @@ -1,55 +0,0 @@ -Nuvoton NAU88L21 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8821" - - - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. - - nuvoton,key_enable: Headset button detection switch. - -Example: - - headset: nau8821@1b { - compatible = "nuvoton,nau8821"; - reg = <0x1b>; - interrupt-parent = <&gpio>; - interrupts = <23 IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,dmic-clk-threshold = 3072000; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml new file mode 100644 index 000000000000..728223d20876 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -0,0 +1,120 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8821.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU88L21 audio codec + +maintainers: + - Seven Lee <wtli(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8821 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: Pull-up JKDET pin. If set then JKDET pin is pull up, + otherwise pull down. + type: boolean + + nuvoton,jkdet-polarity: + description: JKDET pin polarity. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,micbias-voltage: + description: MICBIAS output level select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # VDDA + - 1 # VDDA * 1 + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,vref-impedance: + description: VMID Tie-off impedance select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # open(default) + - 1 # 25KOhms + - 2 # 125KOhms + - 3 # 2.5KOhms + default: 2 + + nuvoton,jack-insert-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,dmic-clk-threshold: + description: DMIC clock speed expected value. Unit is Hz. + $ref: /schemas/types.yaml#/definitions/uint32 + default: 3072000 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1b { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,micbias-voltage = <6>; + nuvoton,vref-impedance = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <0>; + nuvoton,dmic-clk-threshold = <3072000>; + }; + }; -- 2.25.1

3 2

[PATCH v1] ALSA: hda/tas2781: fix 'fw_entry' is used uninitialized
by Shenghao Ding 12 Jul '23

12 Jul '23

Fix 'fw_entry' is used uninitialized. Signed-off-by: Shenghao Ding <13916275206(a)139.com> Changes in v1: | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307070922.doxi1HFc-lkp@intel.com/ --- sound/soc/codecs/tas2781-fmwlib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index eb55abae0d7b..0b08e24ed23a 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1908,7 +1908,7 @@ int tas2781_load_calibration(void *context, char *file_name, { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; struct tasdevice *tasdev = &(tas_priv->tasdevice[i]); - const struct firmware *fw_entry; + const struct firmware *fw_entry = NULL; struct tasdevice_fw *tas_fmw; struct firmware fmw; int offset = 0; -- 2.34.1

1 0

[PATCH v2] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 12 Jul '23

12 Jul '23

This configuration supports JSL boards which implement ALC5650 dual I2S interface codec. Two DAI links are added: AIF1 (on codec side) for headphone and AIF2 for speakers. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/sof_rt5682.c | 81 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 3 files changed, 93 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f472f603ab75..1fe830af2b84 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -475,7 +475,7 @@ endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC if SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_SOF_RT5682_MACH - tristate "SOF with rt5682 codec in I2S Mode" + tristate "SOF with rt5650/rt5682 codec in I2S Mode" depends on I2C && ACPI depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ @@ -485,6 +485,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P + select SND_SOC_RT5645 select SND_SOC_RT5682_I2C select SND_SOC_RT5682S select SND_SOC_DMIC @@ -494,7 +495,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_INTEL_SOF_REALTEK_COMMON help This adds support for ASoC machine driver for SOF platforms - with rt5682 codec. + with rt5650 or rt5682 codec. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7c034d671cf3..53451c1b67f8 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -22,6 +22,7 @@ #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" +#include "../../codecs/rt5645.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" @@ -60,6 +61,7 @@ #define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) #define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) #define SOF_RT1019_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_RT5650_HEADPHONE_CODEC_PRESENT BIT(27) /* Default: MCLK on, MCLK 19.2M, SSP0 */ @@ -305,7 +307,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; - int ret; + int ret, extra; /* need to enable ASRC function for 24MHz mclk rate */ if ((sof_rt5682_quirk & SOF_RT5682_MCLK_EN) && @@ -315,7 +317,16 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) RT5682S_DA_STEREO1_FILTER | RT5682S_AD_STEREO1_FILTER, RT5682S_CLK_SEL_I2S1_ASRC); - else + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(component, + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | RT5682_AD_STEREO1_FILTER, @@ -365,7 +376,12 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); + + if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + extra = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra); + } else + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); @@ -402,6 +418,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_MCLK; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_MCLK; else pll_source = RT5682_PLL1_S_MCLK; @@ -422,6 +440,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, } else { if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_BCLK1; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_BCLK1; else pll_source = RT5682_PLL1_S_BCLK1; @@ -431,6 +451,9 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + pll_id = 0; /* not used in codec driver */ + clk_id = RT5645_SCLK_S_PLL1; } else { pll_id = RT5682_PLL1; clk_id = RT5682_SCLK_S_PLL1; @@ -559,11 +582,30 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "SPOL" }, + { "Right Spk", NULL, "SPOR" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes, + ARRAY_SIZE(rt5650_spk_dapm_routes)); + if (ret) + dev_err(rtd->dev, "fail to add dapm routes, ret=%d\n", ret); + + return ret; +} + static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -614,6 +656,17 @@ static struct snd_soc_dai_link_component rt5682s_component[] = { } }; +static struct snd_soc_dai_link_component rt5650_components[] = { + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif1", + }, + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif2", + } +}; + static struct snd_soc_dai_link_component dmic_component[] = { { .name = "dmic-codec", @@ -652,6 +705,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { links[id].codecs = rt5682s_component; links[id].num_codecs = ARRAY_SIZE(rt5682s_component); + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[0]; + links[id].num_codecs = 1; } else { links[id].codecs = rt5682_component; links[id].num_codecs = ARRAY_SIZE(rt5682_component); @@ -804,6 +860,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].init = max_98390_spk_codec_init; links[id].ops = &max_98390_ops; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[1]; + links[id].num_codecs = 1; + links[id].init = rt5650_spk_init; + links[id].ops = &sof_rt5682_ops; } else { max_98357a_dai_link(&links[id]); } @@ -890,6 +951,12 @@ static int sof_audio_probe(struct platform_device *pdev) /* Detect the headset codec variant */ if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + else if (acpi_dev_present("10EC5650", NULL, -1)) { + sof_rt5682_quirk |= SOF_RT5650_HEADPHONE_CODEC_PRESENT; + + sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", + GFP_KERNEL); + } if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; @@ -1178,6 +1245,14 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(0) | SOF_RT5682_NUM_HDMIDEV(3)), }, + { + .name = "jsl_rt5650", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index f5c7e1bbded0..36ea2bacbe7d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -34,6 +34,11 @@ static const struct snd_soc_acpi_codecs mx98360a_spk = { .codecs = {"MX98360A"} }; +static struct snd_soc_acpi_codecs rt5650_spk = { + .num_codecs = 1, + .codecs = {"10EC5650"} +}; + static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {"10EC5682", "RTL5682"}, @@ -98,6 +103,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .id = "10EC5650", + .drv_name = "jsl_rt5650", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt5650_spk, + .sof_tplg_filename = "sof-jsl-rt5650-dual.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

3 2

[PATCH] ASoC: amd: ps: Fix extraneous error messages
by Mario Limonciello 12 Jul '23

12 Jul '23

On Pink Sardine machines that don't have SdW controllers in use, the property `mipi-sdw-manager-list` won't exist. There is no point in showing an error to a user when this situation is encountered. Furthermore if the machine doesn't have a DMIC connected to the ACP, there may be no platform devices created either. Downgrade the associated message to debug. Fixes: d1351c30ac8a6 ("ASoC: amd: ps: create platform devices based on acp config") Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 5b46dc8573f86..4af3c3665387d 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -257,7 +257,7 @@ static int sdw_amd_scan_controller(struct device *dev) &sdw_manager_bitmap, 1); if (ret) { - dev_err(dev, "Failed to read mipi-sdw-manager-list: %d\n", ret); + dev_dbg(dev, "Failed to read mipi-sdw-manager-list: %d\n", ret); return -EINVAL; } count = hweight32(sdw_manager_bitmap); @@ -641,7 +641,7 @@ static int snd_acp63_probe(struct pci_dev *pci, ret = get_acp63_device_config(val, pci, adata); /* ACP PCI driver probe should be continued even PDM or SoundWire Devices are not found */ if (ret) { - dev_err(&pci->dev, "get acp device config failed:%d\n", ret); + dev_dbg(&pci->dev, "get acp device config failed:%d\n", ret); goto skip_pdev_creation; } ret = create_acp63_platform_devs(pci, adata, addr); base-commit: 8689f4f2ea561dd080118eeb05c0255ac9542905 -- 2.34.1

2 1

[PATCH] ASoC: fsl_sai: Revert "ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode"
by Fabio Estevam 12 Jul '23

12 Jul '23

From: Fabio Estevam <festevam(a)denx.de> This reverts commit ff87d619ac180444db297f043962a5c325ded47b. Andreas reports that on an i.MX8MP-based system where MCLK needs to be used as an input, the MCLK pin is actually an output, despite not having the 'fsl,sai-mclk-direction-output' property present in the devicetree. This is caused by commit ff87d619ac18 ("ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode") that sets FSL_SAI_MCTL_MCLK_EN unconditionally for imx8mm/8mn/8mp/93, causing the MCLK to always be configured as output. FSL_SAI_MCTL_MCLK_EN corresponds to the MOE (MCLK Output Enable) bit of register MCR and the drivers sets it when the 'fsl,sai-mclk-direction-output' devicetree property is present. Revert the commit to allow SAI to use MCLK as input as well. Cc: stable(a)vger.kernel.org Fixes: ff87d619ac18 ("ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode") Reported-by: Andreas Henriksson <andreas(a)fatal.se> Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/fsl/fsl_sai.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5e09f634c61b..54b4bf3744c6 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -507,12 +507,6 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) savediv / 2 - 1); } - if (sai->soc_data->max_register >= FSL_SAI_MCTL) { - /* SAI is in master mode at this point, so enable MCLK */ - regmap_update_bits(sai->regmap, FSL_SAI_MCTL, - FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); - } - return 0; } -- 2.34.1

3 2

[PATCH] ASoC: qcom: q6apm: do not close GPR port before closing graph
by Srinivas Kandagatla 11 Jul '23

11 Jul '23

Closing GPR port before graph close can result in un handled notifications from DSP, this results in spam of errors from GPR driver as there is no one to handle these notification at that point in time. Fix this by closing GPR port after graph close is finished. Fixes: 5477518b8a0e ("ASoC: qdsp6: audioreach: add q6apm support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 7bfac9492ab5..5d44d07acd69 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -511,6 +511,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) switch (hdr->opcode) { case DATA_CMD_RSP_WR_SH_MEM_EP_DATA_BUFFER_DONE_V2: + if (!graph->ar_graph) + break; client_event = APM_CLIENT_EVENT_DATA_WRITE_DONE; mutex_lock(&graph->lock); token = hdr->token & APM_WRITE_TOKEN_MASK; @@ -544,6 +546,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) wake_up(&graph->cmd_wait); break; case DATA_CMD_RSP_RD_SH_MEM_EP_DATA_BUFFER_V2: + if (!graph->ar_graph) + break; client_event = APM_CLIENT_EVENT_DATA_READ_DONE; mutex_lock(&graph->lock); rd_done = data->payload; @@ -649,8 +653,9 @@ int q6apm_graph_close(struct q6apm_graph *graph) { struct audioreach_graph *ar_graph = graph->ar_graph; - gpr_free_port(graph->port); + graph->ar_graph = NULL; kref_put(&ar_graph->refcount, q6apm_put_audioreach_graph); + gpr_free_port(graph->port); kfree(graph); return 0; -- 2.25.1

2 1

[PATCH 0/8] ASoC/soundwire/qdsp6/wcd: fix leaks and probe deferral
by Johan Hovold 11 Jul '23

11 Jul '23

I've been hitting a race during boot which breaks probe of the sound card on the Lenovo ThinkPad X13s as I've previously reported here: https://lore.kernel.org/all/ZIHMMFtuDtvdpFAZ@hovoldconsulting.com/ The immediate issue appeared to be a probe deferral that was turned into a hard failure, but addressing that in itself only made things worse as it exposed further bugs. I was hoping someone more familiar with the code in question would look into this, but as this affects users of the X13s and breaks audio on my machine every fifth boot or so, I decided to investigate it myself. As expected, the Qualcomm codec drivers are broken and specifically leak resources on component remove, which in turn breaks sound card probe deferrals. The source of the deferral itself appears to be legitimate and was simply due to some audio component not yet having been registered due to random changes in timing during boot. These issues can most easily be reproduced by simply blacklisting the q6apm_dai module and loading it manually after boot. The sound card probe deferral also exposes a bug in the soundwire subsystem, which uses completion structures for signalling that a device has been enumerated on the bus and initialised. The way this is implemented prevents reprobed codec drivers from learning that the soundwire devices are still attached, which causes probe to fail. Included are also two patches that suppresses error messages on component probe deferral to avoid spamming the logs during boot. These patches should preferably all go through the ASoC tree even if merging the soundwire fix separately also works. Note the ASoC tree already has the following related fixes: https://lore.kernel.org/lkml/20230630120318.6571-1-johan+linaro@kernel.org/ https://lore.kernel.org/lkml/20230630142717.5314-1-johan+linaro@kernel.org/ https://lore.kernel.org/lkml/20230701094723.29379-1-johan+linaro@kernel.org/ https://lore.kernel.org/lkml/20230703124701.11734-1-johan+linaro@kernel.org/ Johan Johan Hovold (8): soundwire: fix enumeration completion ASoC: qdsp6: audioreach: fix topology probe deferral ASoC: codecs: wcd938x: fix missing clsh ctrl error handling ASoC: codecs: wcd938x: fix resource leaks on component remove ASoC: codecs: wcd934x: fix resource leaks on component remove ASoC: codecs: wcd-mbhc-v2: fix resource leaks on component remove ASoC: topology: suppress probe deferral errors ASoC: core: suppress probe deferral errors drivers/soundwire/bus.c | 8 ++--- sound/soc/codecs/wcd-mbhc-v2.c | 57 ++++++++++++++++++++++--------- sound/soc/codecs/wcd934x.c | 12 +++++++ sound/soc/codecs/wcd938x.c | 59 +++++++++++++++++++++++++++++---- sound/soc/qcom/qdsp6/topology.c | 4 +-- sound/soc/soc-core.c | 6 ++-- sound/soc/soc-topology.c | 10 ++++-- 7 files changed, 122 insertions(+), 34 deletions(-) -- 2.39.3

7 22

[PATCH] ASoC: codecs: aw88395: Add of_match_table
by Konrad Dybcio 11 Jul '23

11 Jul '23

The i2c_match_id entry is not enough to get the device probing with DT. Add an of match table in accordance with the bindings. Fixes: 62fc25fbab5f ("ASoC: codecs: Add i2c and codec registration for aw88395 and their associated operation functions") Signed-off-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> --- sound/soc/codecs/aw88395/aw88395.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/aw88395/aw88395.c b/sound/soc/codecs/aw88395/aw88395.c index 9dcd75dd799a..5ff01d3c31c8 100644 --- a/sound/soc/codecs/aw88395/aw88395.c +++ b/sound/soc/codecs/aw88395/aw88395.c @@ -566,9 +566,18 @@ static const struct i2c_device_id aw88395_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aw88395_i2c_id); +#ifdef CONFIG_OF +static const struct of_device_id aw88395_of_match[] = { + { .compatible = "awinic,aw88395" }, + { }, +}; +MODULE_DEVICE_TABLE(of, aw88395_of_match); +#endif + static struct i2c_driver aw88395_i2c_driver = { .driver = { .name = AW88395_I2C_NAME, + .of_match_table = of_match_ptr(aw88395_of_match), }, .probe = aw88395_i2c_probe, .id_table = aw88395_i2c_id, --- base-commit: 8e4b7f2f3d6071665b1dfd70786229c8a5d6c256 change-id: 20230711-topic-aw88395_of_tbl-68b16c5860a5 Best regards, -- Konrad Dybcio <konrad.dybcio(a)linaro.org>

1 0

[PATCH v3 00/14] spi: Header and core clean up and refactoring
by Andy Shevchenko 11 Jul '23

11 Jul '23

Various cleanups and refactorings of the SPI header and core parts united in a single series. It also touches drivers under SPI subsystem folder on the pure renaming purposes of some constants. No functional change intended (with some subtle shortcuts which are explained in the respective commit messages). Changelog v3: - dropped controversial used to be patches 3,4,8,10 (Mark) - amended many commit messages (Mark) - added tag to patch 1 (AngeloGioacchino) - split used to be patch 2 to patches 2 & 3 for better review (Mark) - rewritten used to be patch 5 to patches 4 & 5 (Mark, Sebastian) - added new patch 7 - fixed typos and added tag to patch 12 (Serge) v2: 20230710154932.68377-1-andriy.shevchenko(a)linux.intel.com Changelog v2: - added new patches 3,4,5,10,13,14 - massaged comment and kernel doc in patch 9 - split used to be patch 4 to patches 11,12 - covered a few things in SPI core in patch 15 - amended commit message for above (Mark) - reshuffled patches in the series for better logical grouping Andy Shevchenko (14): spi: Remove unneeded OF node NULL checks spi: Deduplicate IDR allocation code in spi_register_controller() spi: Refactor bus number assigning code in spi_register_controller() spi: Remove code duplication in spi_add_device*() spi: Kill spi_add_device_locked() spi: Use sysfs_emit() to instead of s*printf() spi: Use BITS_TO_BYTES() spi: Sort headers alphabetically spi: Use struct_size() helper spi: Get rid of old SPI_MASTER_NO_TX & SPI_MASTER_NO_RX spi: Get rid of old SPI_MASTER_MUST_TX & SPI_MASTER_MUST_RX spi: Rename SPI_MASTER_GPIO_SS to SPI_CONTROLLER_GPIO_SS spi: Convert to SPI_CONTROLLER_HALF_DUPLEX spi: Fix spelling typos and acronyms capitalization drivers/spi/spi-amd.c | 2 +- drivers/spi/spi-at91-usart.c | 2 +- drivers/spi/spi-ath79.c | 2 +- drivers/spi/spi-atmel.c | 4 +- drivers/spi/spi-bitbang-txrx.h | 16 +-- drivers/spi/spi-bitbang.c | 8 +- drivers/spi/spi-cavium-thunderx.c | 2 +- drivers/spi/spi-davinci.c | 2 +- drivers/spi/spi-dw-core.c | 2 +- drivers/spi/spi-falcon.c | 2 +- drivers/spi/spi-fsl-lpspi.c | 2 +- drivers/spi/spi-gpio.c | 10 +- drivers/spi/spi-imx.c | 2 +- drivers/spi/spi-lp8841-rtc.c | 10 +- drivers/spi/spi-meson-spicc.c | 2 +- drivers/spi/spi-mt65xx.c | 2 +- drivers/spi/spi-mxs.c | 2 +- drivers/spi/spi-omap-uwire.c | 2 +- drivers/spi/spi-orion.c | 2 +- drivers/spi/spi-pci1xxxx.c | 2 +- drivers/spi/spi-pic32-sqi.c | 2 +- drivers/spi/spi-pic32.c | 2 +- drivers/spi/spi-qcom-qspi.c | 2 +- drivers/spi/spi-rb4xx.c | 2 +- drivers/spi/spi-rockchip-sfc.c | 2 +- drivers/spi/spi-rockchip.c | 2 +- drivers/spi/spi-sifive.c | 2 +- drivers/spi/spi-slave-mt27xx.c | 2 +- drivers/spi/spi-sprd-adi.c | 2 +- drivers/spi/spi-stm32.c | 2 +- drivers/spi/spi-ti-qspi.c | 2 +- drivers/spi/spi-xcomm.c | 2 +- drivers/spi/spi-xtensa-xtfpga.c | 2 +- drivers/spi/spi.c | 211 +++++++++++++----------------- include/linux/spi/spi.h | 173 ++++++++++++------------ include/trace/events/spi.h | 2 +- 36 files changed, 238 insertions(+), 252 deletions(-) -- 2.40.0.1.gaa8946217a0b

2 15

[PATCH 1/2] ALSA: emu10k1: make E-MU dock monitoring interrupt-driven
by Oswald Buddenhagen 11 Jul '23

11 Jul '23

... instead of using a one-second polling timer. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- include/sound/emu10k1.h | 4 +-- sound/pci/emu10k1/emu10k1.c | 8 +---- sound/pci/emu10k1/emu10k1_main.c | 51 ++++++++++++++++++-------------- sound/pci/emu10k1/io.c | 2 ++ sound/pci/emu10k1/irq.c | 7 +++++ 5 files changed, 40 insertions(+), 32 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 386a5f3be3e0..43c097952c3c 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1678,8 +1678,7 @@ struct snd_emu1010 { unsigned int clock_fallback; unsigned int optical_in; /* 0:SPDIF, 1:ADAT */ unsigned int optical_out; /* 0:SPDIF, 1:ADAT */ - struct delayed_work firmware_work; - u32 last_reg; + struct work_struct firmware_work; }; struct snd_emu10k1 { @@ -1761,6 +1760,7 @@ struct snd_emu10k1 { void (*capture_efx_interrupt)(struct snd_emu10k1 *emu, unsigned int status); void (*spdif_interrupt)(struct snd_emu10k1 *emu, unsigned int status); void (*dsp_interrupt)(struct snd_emu10k1 *emu); + void (*gpio_interrupt)(struct snd_emu10k1 *emu); void (*p16v_interrupt)(struct snd_emu10k1 *emu); struct snd_pcm_substream *pcm_capture_substream; diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 23adace1b969..1a13c086e86c 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -176,9 +176,6 @@ static int snd_card_emu10k1_probe(struct pci_dev *pci, if (err < 0) return err; - if (emu->card_capabilities->emu_model) - schedule_delayed_work(&emu->emu1010.firmware_work, 0); - pci_set_drvdata(pci, card); dev++; return 0; @@ -194,7 +191,7 @@ static int snd_emu10k1_suspend(struct device *dev) emu->suspend = 1; - cancel_delayed_work_sync(&emu->emu1010.firmware_work); + cancel_work_sync(&emu->emu1010.firmware_work); snd_ac97_suspend(emu->ac97); @@ -224,9 +221,6 @@ static int snd_emu10k1_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); - if (emu->card_capabilities->emu_model) - schedule_delayed_work(&emu->emu1010.firmware_work, 0); - return 0; } diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 58ed72de6403..661164dbf547 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -391,7 +391,10 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) } #endif - snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE); + if (emu->card_capabilities->emu_model) + snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE | INTE_A_GPIOENABLE); + else + snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE); } int snd_emu10k1_done(struct snd_emu10k1 *emu) @@ -745,63 +748,61 @@ static void emu1010_firmware_work(struct work_struct *work) int err; emu = container_of(work, struct snd_emu10k1, - emu1010.firmware_work.work); + emu1010.firmware_work); if (emu->card->shutdown) return; #ifdef CONFIG_PM_SLEEP if (emu->suspend) return; #endif - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp); /* IRQ Status */ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg); /* OPTIONS: Which cards are attached to the EMU */ if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { /* Audio Dock attached */ /* Return to Audio Dock programming mode */ dev_info(emu->card->dev, "emu1010: Loading Audio Dock Firmware\n"); snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK); err = snd_emu1010_load_firmware(emu, 1, &emu->dock_fw); if (err < 0) - goto next; - + return; snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0); - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp); - dev_info(emu->card->dev, - "emu1010: EMU_HANA+DOCK_IRQ_STATUS = 0x%x\n", tmp); - /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &tmp); dev_info(emu->card->dev, "emu1010: EMU_HANA+DOCK_ID = 0x%x\n", tmp); if ((tmp & 0x1f) != 0x15) { /* FPGA failed to be programmed */ dev_info(emu->card->dev, "emu1010: Loading Audio Dock Firmware file failed, reg = 0x%x\n", tmp); - goto next; + return; } dev_info(emu->card->dev, "emu1010: Audio Dock Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); dev_info(emu->card->dev, "Audio Dock ver: %u.%u\n", tmp, tmp2); /* Sync clocking between 1010 and Dock */ /* Allow DLL to settle */ msleep(10); /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); - } else if (!reg && emu->emu1010.last_reg) { + } +} + +static void emu1010_interrupt(struct snd_emu10k1 *emu) +{ + u32 sts; + + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &sts); + if (sts & EMU_HANA_IRQ_DOCK_LOST) { /* Audio Dock removed */ dev_info(emu->card->dev, "emu1010: Audio Dock detached\n"); /* The hardware auto-mutes all, so we unmute again */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); + } else if (sts & EMU_HANA_IRQ_DOCK) { + schedule_work(&emu->emu1010.firmware_work); } - - next: - emu->emu1010.last_reg = reg; - if (!emu->card->shutdown) - schedule_delayed_work(&emu->emu1010.firmware_work, - msecs_to_jiffies(1000)); } /* @@ -870,6 +871,8 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg); dev_info(emu->card->dev, "emu1010: Card options = 0x%x\n", reg); + if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) + schedule_work(&emu->emu1010.firmware_work); if (emu->card_capabilities->no_adat) { emu->emu1010.optical_in = 0; /* IN_SPDIF */ emu->emu1010.optical_out = 0; /* OUT_SPDIF */ @@ -895,10 +898,12 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) /* MIDI routing */ snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, EMU_HANA_MIDI_INA_FROM_HAMOA | EMU_HANA_MIDI_INB_FROM_DOCK2); snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, EMU_HANA_MIDI_OUT_DOCK2 | EMU_HANA_MIDI_OUT_SYNC2); - /* IRQ Enable: All on */ - /* snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x0f); */ - /* IRQ Enable: All off */ - snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00); + + emu->gpio_interrupt = emu1010_interrupt; + // Note: The Audigy INTE is set later + snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, + EMU_HANA_IRQ_DOCK | EMU_HANA_IRQ_DOCK_LOST); + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg); // Clear pending IRQs emu->emu1010.clock_source = 1; /* 48000 */ emu->emu1010.clock_fallback = 1; /* 48000 */ @@ -938,7 +943,7 @@ static void snd_emu10k1_free(struct snd_card *card) /* Disable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); } - cancel_delayed_work_sync(&emu->emu1010.firmware_work); + cancel_work_sync(&emu->emu1010.firmware_work); release_firmware(emu->firmware); release_firmware(emu->dock_fw); snd_util_memhdr_free(emu->memhdr); @@ -1517,7 +1522,7 @@ int snd_emu10k1_create(struct snd_card *card, emu->irq = -1; emu->synth = NULL; emu->get_synth_voice = NULL; - INIT_DELAYED_WORK(&emu->emu1010.firmware_work, emu1010_firmware_work); + INIT_WORK(&emu->emu1010.firmware_work, emu1010_firmware_work); /* read revision & serial */ emu->revision = pci->revision; pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial); diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index a0d66ce3ee83..c695cb863e5e 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -302,6 +302,8 @@ static void snd_emu1010_fpga_read_locked(struct snd_emu10k1 *emu, u32 reg, u32 * { // The higest input pin is used as the designated interrupt trigger, // so it needs to be masked out. + // But note that any other input pin change will also cause an IRQ, + // so using this function often causes an IRQ as a side effect. u32 mask = emu->card_capabilities->ca0108_chip ? 0x1f : 0x7f; if (snd_BUG_ON(reg > 0x3f)) return; diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c index a813ef8c2f8d..8573248dd799 100644 --- a/sound/pci/emu10k1/irq.c +++ b/sound/pci/emu10k1/irq.c @@ -149,6 +149,13 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id) outl(0, emu->port + INTE2); status &= ~IPR_P16V; } + if (status & IPR_A_GPIO) { + if (emu->gpio_interrupt) + emu->gpio_interrupt(emu); + else + snd_emu10k1_intr_disable(emu, INTE_A_GPIOENABLE); + status &= ~IPR_A_GPIO; + } if (status) { dev_err(emu->card->dev, -- 2.40.0.152.g15d061e6df

2 7

ALC4082 - ASRock Z790 PG-ITX/TB4
by GitHub issues - edited 11 Jul '23

11 Jul '23

alsa-project/alsa-ucm-conf issue #334 was edited from Shmob1: I have an [ASRock Z790 PG-ITX/TB4](https://pg.asrock.com/MB/Intel/Z790%20PG-ITXTB4/index.asp) which has the `ALC4082` Realtek chipset connected over USB, that has been mentioned in a number of issues, most recently #229. It seems the configuration changes in that issue (notably, https://github.com/hwchong/alsa-ucm-conf/commit/3ec7e6e2a61240764b77c30892f…) work on my board as well, with a small modification. Unlike the Taichi boards, my Realtek audio driver is at `USB26ce:0a08`, not `USB26ce:0a06`. Making the necessary regex changes in `ALC4080-HiFi.conf` and `USB-Audio.conf` enables functionality on my board. I am happy to make a PR to merge these changes if pertinent. Thanks for the great work! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/334 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock Z790 PG-ITX/TB4
by GitHub issues - opened 11 Jul '23

11 Jul '23

alsa-project/alsa-ucm-conf issue #334 was opened from Shmob1: I have an [ASRock Z790 PG-ITX/TB4](https://pg.asrock.com/MB/Intel/Z790%20PG-ITXTB4/index.asp) which has the `ALC4082` Realtek chipset connected over USB, that has been mentioned in a number of issues, most recently #229. It seems the configuration changes in that issue (notably, https://github.com/hwchong/alsa-ucm-conf/commit/3ec7e6e2a61240764b77c30892f…) work on my board as well, with a small modification. Unlike the Taichi boards, my Realtek audio driver is at `USB26ce:0a08`, not `USB26ce:0a06`. Making the necessary regex changes in `ALC4080-HiFi.conf` and `USB-Audio.conf` enables functionality on my board. I am happy to make a PR to merge these changes if required. Thanks for the great work! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/334 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 1/3] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 10 Jul '23

10 Jul '23

Integrate tas2781 configs for Lenovo Laptops. All of the tas2781s in the laptop will be aggregated as one audio device. The code support realtek as the primary codec. Rename "struct cs35l41_dev_name" to "struct scodec_dev_name" for all other side codecs instead of the certain one. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v2: - simplify the check of vendor id with Lenovo - ThinkPad is one of Lenovo's brands, they suggested me to use ALC269_FIXUP_THINKPAD_ACPI. - Add comments on ACARD_SINGLE_RANGE_EXT_TLV - Add the range check for tas_priv->tasdevice[] in tas2781_acpi_get_i2c_res. - remove acpi_subsystem_id - Issue in Laptop 0x17aa38be ACPI talbe caused codec->bus->pci->subsystem_device is not equal to (codec->core.subsystem_id & 0xffff) in snd_hda_pick_fixup. The former is 0x3802 and the latter is 0x38be leads to getting the wrong fixup_id and enter into the wrong entry. Although, this issue has been raised to the laptop manufacturer, but the ACPI table is locked, cannot be changed any more. Correct the wrong entry in the code. - Rename "struct cs35l41_dev_name" to "struct scodec_dev_name" for all other side codecs instead of one certain codec. - Ignore the checkpatch complaints in alc269_fixup_tbl - Drop the hunk which is irrelevant with my code in alc_fixup_headset_mode_alc255_no_hp_mic - Add tiwai(a)suse.de into Cc list - remove useless index - combine ALC287_FIXUP_TAS2781_I2C_2 and ALC287_FIXUP_TAS2781_I2C_4 together as ALC287_FIXUP_TAS2781_I2C, The code view all the tas2781s in the laptop as one instance. - delete the white space at the end of the line in alc_fixup_headset_mode_alc255_no_hp_mic --- sound/pci/hda/patch_realtek.c | 104 +++++++++++++++++++++++++++++++++- 1 file changed, 101 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e2f8b608de82..d730497af57d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6705,7 +6705,7 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ } } -struct cs35l41_dev_name { +struct scodec_dev_name { const char *bus; const char *hid; int index; @@ -6714,7 +6714,7 @@ struct cs35l41_dev_name { /* match the device name in a slightly relaxed manner */ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) { - struct cs35l41_dev_name *p = data; + struct scodec_dev_name *p = data; const char *d = dev_name(dev); int n = strlen(p->bus); char tmp[32]; @@ -6730,12 +6730,32 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +static int comp_match_tas2781_dev_name(struct device *dev, + void *data) +{ + struct scodec_dev_name *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->bus, n)) + return 0; + /* skip the bus number */ + if (isdigit(d[n])) + n++; + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); + + return !strcmp(d + n, tmp); +} + static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct cs35l41_dev_name *rec; + struct scodec_dev_name *rec; int ret, i; switch (action) { @@ -6763,6 +6783,41 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } } +static void tas2781_generic_fixup(struct hda_codec *cdc, int action, + const char *bus, const char *hid) +{ + struct device *dev = hda_codec_dev(cdc); + struct alc_spec *spec = cdc->spec; + struct scodec_dev_name *rec; + int ret; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); + if (!rec) + return; + rec->bus = bus; + rec->hid = hid; + rec->index = 0; + spec->comps[0].codec = cdc; + component_match_add(dev, &spec->match, + comp_match_tas2781_dev_name, rec); + ret = component_master_add_with_match(dev, &comp_master_ops, + spec->match); + if (ret) + codec_err(cdc, + "Fail to register component aggregator %d\n", + ret); + else + spec->gen.pcm_playback_hook = + comp_generic_playback_hook; + break; + case HDA_FIXUP_ACT_FREE: + component_master_del(dev, &comp_master_ops); + break; + } +} + static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); @@ -6790,6 +6845,12 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); } +static void tas2781_fixup_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7213,6 +7274,7 @@ enum { ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, + ALC287_FIXUP_TAS2781_I2C, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9283,6 +9345,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC287_FIXUP_TAS2781_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9846,6 +9914,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -10761,6 +10843,22 @@ static int patch_alc269(struct hda_codec *codec) codec->fixup_id = HDA_FIXUP_ID_NOT_SET; } + /* FIXME: Issue in Laptop 0x17aa38be ACPI talbe caused + * codec->bus->pci->subsystem_device is not equal to + * (codec->core.subsystem_id & 0xffff) in snd_hda_pick_fixup. + * The former is 0x3802 and the latter is 0x38be leads to getting the + * wrong fixup_id and enter into the wrong entry. Although, this issue + * has been raised to the laptop manufacturer, but the ACPI table is + * locked, cannot be changed any more. Correct the wrong entry in the + * code. + */ + if (codec->fixup_id == ALC287_FIXUP_YOGA7_14ITL_SPEAKERS && + codec->core.vendor_id == 0x10ec0287 && + codec->core.subsystem_id == 0x17aa38be) { + codec_dbg(codec, "Clear wrong fixup for 17aa38be\n"); + codec->fixup_id = ALC287_FIXUP_TAS2781_I2C; + } + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true); snd_hda_pick_pin_fixup(codec, alc269_fallback_pin_fixup_tbl, alc269_fixups, false); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, -- 2.34.1

2 4

[PATCH v1 0/8] spi: Header and core clean up and refactoring
by Andy Shevchenko 10 Jul '23

10 Jul '23

Various cleanups and refactorings of the SPI header and core parts united in a single series. Patches 1 & 2, 5 & 6 & 8 are dependent inside each group. No functional change intended. Andy Shevchenko (8): spi: Remove unneeded OF node NULL checks spi: Drop duplicate IDR allocation code in spi_register_controller() spi: Use sysfs_emit() to instead of s*printf() spi: Get rid of old SPI_MASTER_NO_.X and SPI_MASTER_MUST_.X spi: Sort headers alphabetically spi: Clean up headers spi: Fix spelling typos and acronyms capitalization spi: Use struct_size() helper drivers/spi/spi-at91-usart.c | 2 +- drivers/spi/spi-atmel.c | 2 +- drivers/spi/spi-bitbang-txrx.h | 16 +-- drivers/spi/spi-bitbang.c | 2 +- drivers/spi/spi-davinci.c | 2 +- drivers/spi/spi-fsl-lpspi.c | 2 +- drivers/spi/spi-gpio.c | 8 +- drivers/spi/spi-lp8841-rtc.c | 8 +- drivers/spi/spi-meson-spicc.c | 2 +- drivers/spi/spi-mt65xx.c | 2 +- drivers/spi/spi-pci1xxxx.c | 2 +- drivers/spi/spi-pic32.c | 2 +- drivers/spi/spi-rb4xx.c | 2 +- drivers/spi/spi-slave-mt27xx.c | 2 +- drivers/spi/spi-stm32.c | 2 +- drivers/spi/spi-xtensa-xtfpga.c | 2 +- drivers/spi/spi.c | 102 ++++++++--------- include/linux/spi/spi.h | 188 ++++++++++++++++++-------------- 18 files changed, 183 insertions(+), 165 deletions(-) -- 2.40.0.1.gaa8946217a0b

2 18

[PATCH] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 10 Jul '23

10 Jul '23

This configuration supports JSL boards which implement ALC5650 dual I2S interface codec. Two DAI links are added: AIF1 (on codec side) for headphone and AIF2 for speakers. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 81 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 2 files changed, 90 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7c034d671cf3..53451c1b67f8 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -22,6 +22,7 @@ #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" +#include "../../codecs/rt5645.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" @@ -60,6 +61,7 @@ #define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) #define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) #define SOF_RT1019_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_RT5650_HEADPHONE_CODEC_PRESENT BIT(27) /* Default: MCLK on, MCLK 19.2M, SSP0 */ @@ -305,7 +307,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; - int ret; + int ret, extra; /* need to enable ASRC function for 24MHz mclk rate */ if ((sof_rt5682_quirk & SOF_RT5682_MCLK_EN) && @@ -315,7 +317,16 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) RT5682S_DA_STEREO1_FILTER | RT5682S_AD_STEREO1_FILTER, RT5682S_CLK_SEL_I2S1_ASRC); - else + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(component, + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | RT5682_AD_STEREO1_FILTER, @@ -365,7 +376,12 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); + + if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + extra = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra); + } else + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); @@ -402,6 +418,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_MCLK; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_MCLK; else pll_source = RT5682_PLL1_S_MCLK; @@ -422,6 +440,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, } else { if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_BCLK1; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_BCLK1; else pll_source = RT5682_PLL1_S_BCLK1; @@ -431,6 +451,9 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + pll_id = 0; /* not used in codec driver */ + clk_id = RT5645_SCLK_S_PLL1; } else { pll_id = RT5682_PLL1; clk_id = RT5682_SCLK_S_PLL1; @@ -559,11 +582,30 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "SPOL" }, + { "Right Spk", NULL, "SPOR" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes, + ARRAY_SIZE(rt5650_spk_dapm_routes)); + if (ret) + dev_err(rtd->dev, "fail to add dapm routes, ret=%d\n", ret); + + return ret; +} + static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -614,6 +656,17 @@ static struct snd_soc_dai_link_component rt5682s_component[] = { } }; +static struct snd_soc_dai_link_component rt5650_components[] = { + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif1", + }, + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif2", + } +}; + static struct snd_soc_dai_link_component dmic_component[] = { { .name = "dmic-codec", @@ -652,6 +705,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { links[id].codecs = rt5682s_component; links[id].num_codecs = ARRAY_SIZE(rt5682s_component); + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[0]; + links[id].num_codecs = 1; } else { links[id].codecs = rt5682_component; links[id].num_codecs = ARRAY_SIZE(rt5682_component); @@ -804,6 +860,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].init = max_98390_spk_codec_init; links[id].ops = &max_98390_ops; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[1]; + links[id].num_codecs = 1; + links[id].init = rt5650_spk_init; + links[id].ops = &sof_rt5682_ops; } else { max_98357a_dai_link(&links[id]); } @@ -890,6 +951,12 @@ static int sof_audio_probe(struct platform_device *pdev) /* Detect the headset codec variant */ if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + else if (acpi_dev_present("10EC5650", NULL, -1)) { + sof_rt5682_quirk |= SOF_RT5650_HEADPHONE_CODEC_PRESENT; + + sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", + GFP_KERNEL); + } if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; @@ -1178,6 +1245,14 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(0) | SOF_RT5682_NUM_HDMIDEV(3)), }, + { + .name = "jsl_rt5650", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index f5c7e1bbded0..36ea2bacbe7d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -34,6 +34,11 @@ static const struct snd_soc_acpi_codecs mx98360a_spk = { .codecs = {"MX98360A"} }; +static struct snd_soc_acpi_codecs rt5650_spk = { + .num_codecs = 1, + .codecs = {"10EC5650"} +}; + static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {"10EC5682", "RTL5682"}, @@ -98,6 +103,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .id = "10EC5650", + .drv_name = "jsl_rt5650", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt5650_spk, + .sof_tplg_filename = "sof-jsl-rt5650-dual.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

1 0

[PATCH v2] soundwire: amd: Fix a check for errors in probe()
by Dan Carpenter 10 Jul '23

10 Jul '23

This code has two problems: 1) The devm_ioremap() function returns NULL, not error pointers. 2) It's checking the wrong variable. ->mmio instead of ->acp_mmio. Fixes: a673a8dfc214 ("soundwire: amd: Add support for AMD Manager driver") Suggested-by: "Mukunda,Vijendar" <vijendar.mukunda(a)amd.com> Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- v2: Change the variable. My original fix would have prevented the driver from loading so many thanks to Vijendar for pointing that out. drivers/soundwire/amd_manager.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 08aeb7ed00e1..3a99f6dcdfaf 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -910,9 +910,9 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) return -ENOMEM; amd_manager->acp_mmio = devm_ioremap(dev, res->start, resource_size(res)); - if (IS_ERR(amd_manager->mmio)) { + if (!amd_manager->acp_mmio) { dev_err(dev, "mmio not found\n"); - return PTR_ERR(amd_manager->mmio); + return -ENOMEM; } amd_manager->instance = pdata->instance; amd_manager->mmio = amd_manager->acp_mmio + -- 2.39.2

2 1

[PATCH v2] sound: seq: fix typo in system_2p_ev_to_ump_midi1()
by Minjie Du 10 Jul '23

10 Jul '23

Fix data->system.parm2 typo. Signed-off-by: Minjie Du <duminjie(a)vivo.com> --- sound/core/seq/seq_ump_convert.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index eb1d86ff6..7cc84e137 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -714,7 +714,7 @@ static int system_2p_ev_to_ump_midi1(const struct snd_seq_event *event, { data->system.status = status; data->system.parm1 = (event->data.control.value >> 7) & 0x7f; - data->system.parm1 = event->data.control.value & 0x7f; + data->system.parm2 = event->data.control.value & 0x7f; return 1; } -- 2.39.0

3 2

[PATCH] ASoC: codecs: wcd938x: fix dB range for HPHL and HPHR
by Srinivas Kandagatla 10 Jul '23

10 Jul '23

dB range for HPHL and HPHR gains are from +6dB to -30dB in steps of 1.5dB with register values range from 0 to 24. Current code maps these dB ranges incorrectly, fix them to allow proper volume setting. Fixes: e8ba1e05bdc0("ASoC: codecs: wcd938x: add basic controls") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd938x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index faa15a5ed2c8..3a3360711f8f 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -210,7 +210,7 @@ struct wcd938x_priv { }; static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(ear_pa_gain, 600, -1800); -static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(line_gain, 600, -3000); +static const DECLARE_TLV_DB_SCALE(line_gain, -3000, 150, -3000); static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(analog_gain, 0, 3000); struct wcd938x_mbhc_zdet_param { @@ -2655,8 +2655,8 @@ static const struct snd_kcontrol_new wcd938x_snd_controls[] = { wcd938x_get_swr_port, wcd938x_set_swr_port), SOC_SINGLE_EXT("DSD_R Switch", WCD938X_DSD_R, 0, 1, 0, wcd938x_get_swr_port, wcd938x_set_swr_port), - SOC_SINGLE_TLV("HPHL Volume", WCD938X_HPH_L_EN, 0, 0x18, 0, line_gain), - SOC_SINGLE_TLV("HPHR Volume", WCD938X_HPH_R_EN, 0, 0x18, 0, line_gain), + SOC_SINGLE_TLV("HPHL Volume", WCD938X_HPH_L_EN, 0, 0x18, 1, line_gain), + SOC_SINGLE_TLV("HPHR Volume", WCD938X_HPH_R_EN, 0, 0x18, 1, line_gain), WCD938X_EAR_PA_GAIN_TLV("EAR_PA Volume", WCD938X_ANA_EAR_COMPANDER_CTL, 2, 0x10, 0, ear_pa_gain), SOC_SINGLE_EXT("ADC1 Switch", WCD938X_ADC1, 1, 1, 0, -- 2.25.1

4 12

[PATCH v2 0/4] soundwire: qcom: stablity fixes
by Srinivas Kandagatla 10 Jul '23

10 Jul '23

During x13s audio testing we hit few corner cases due to issues in codec drivers and some obvious code bugs. Here are the fixes for those issues, mostly the issues are around devices loosing the sync in between runtime pm suspend resume path. With codec fixes along with these fixes, audio on x13s is pretty stable. Thanks, Srini Changes since v1: - dropped runtime pm changes patch as unable to reproduced it anymore - fixed clk stop flag as suggested by Pierre - rebased on top of linux-next Srinivas Kandagatla (4): soundwire: qcom: update status correctly with mask soundwire: qcom: wait for fifo to be empty before suspend soundwire: qcom: add software workaround for bus clash interrupt assertion soundwire: qcom: set clk stop need reset flag at runtime drivers/soundwire/qcom.c | 100 ++++++++++++++++++++++++++------------- 1 file changed, 68 insertions(+), 32 deletions(-) -- 2.21.0

3 13

on (not) breaking user space
by Oswald Buddenhagen 09 Jul '23

09 Jul '23

hi linus, the alsa maintainers seem to be applying a reading of "we don't break user space" that i find counter-productive. so i would appreciate you clarifying your position on that matter (once more ...) to be clear, we're not arguing over changing any abi, but simply exercising one somewhat more than before. the claim is that this would likely have unacceptable consequences. i disagree. context: > Due to hardware constraints, changing the clock multiplier (CM) > changes the available audio ports and the number of available > channels. [...] Therefore, we do dynamic reconfiguration (DR) of the > mixer in response to changing the CM. arguments: > DR is somewhat controversial, as it has the potential to crash poorly > programmed applications. > But that in itself isn't a very convincing argument against it, as by > that logic we'd have to ban all hot-plugging. (1) > Such crashes would also not really qualify as regressions, as the > D.A.S. mode is a new opt-in feature, and therefore no previously > stable setups would be impacted. (2) the full text is at https://lore.kernel.org/all/20230630144542.664190-7-oswald.buddenhagen@gmx.… the preceding discussion is at https://lore.kernel.org/all/87v8fren1k.wl-tiwai@suse.de/T/#u regards, ossi

1 0

[Feature Request] New hdajackretask ncurses TUI
by GitHub issues - opened 09 Jul '23

09 Jul '23

alsa-project/alsa-tools issue #16 was opened from HalanoSiblee: Hi Why i need gtk GUI when I do such a simple retaskable jack input Please make hdajackretask with ncurses like alsamixer to work in tty and terminal's at this point it will be so useful less bloat ,and blazingly fast Thanks. Issue URL : https://github.com/alsa-project/alsa-tools/issues/16 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH 1/2] ALSA: pcmtest: Convert to platform remove callback returning void
by Uwe Kleine-König 07 Jul '23

07 Jul '23

The .remove() callback for a platform driver returns an int which makes many driver authors wrongly assume it's possible to do error handling by returning an error code. However the value returned is ignored (apart from emitting a warning) and this typically results in resource leaks. To improve here there is a quest to make the remove callback return void. In the first step of this quest all drivers are converted to .remove_new() which already returns void. Eventually after all drivers are converted, .remove_new() is renamed to .remove(). Trivially convert this driver from always returning zero in the remove callback to the void returning variant. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/drivers/pcmtest.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 2ae912a64d16..1fff85feaf82 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -576,10 +576,9 @@ static int pcmtst_probe(struct platform_device *pdev) return 0; } -static int pdev_remove(struct platform_device *dev) +static void pdev_remove(struct platform_device *dev) { snd_pcmtst_free(pcmtst); - return 0; } static struct platform_device pcmtst_pdev = { @@ -589,7 +588,7 @@ static struct platform_device pcmtst_pdev = { static struct platform_driver pcmtst_pdrv = { .probe = pcmtst_probe, - .remove = pdev_remove, + .remove_new = pdev_remove, .driver = { .name = "pcmtest", }, base-commit: 5133c9e51de41bfa902153888e11add3342ede18 -- 2.39.2

3 5

[PATCH] dt-bindings: sound: Update maintainer email id
by Rohit kumar 07 Jul '23

07 Jul '23

Updated my mail id to latest quicinc id. Signed-off-by: Rohit kumar <quic_rohkumar(a)quicinc.com> --- .../devicetree/bindings/sound/google,sc7180-trogdor.yaml | 2 +- Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 666a95ac22c8..ba5b7728cf33 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Google SC7180-Trogdor ASoC sound card driver maintainers: - - Rohit kumar <rohitkr(a)codeaurora.org> + - Rohit kumar <quic_rohkumar(a)quicinc.com> - Cheng-Yi Chiang <cychiang(a)chromium.org> description: diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml index 6cc8f86c7531..3a559bd07a79 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -8,7 +8,7 @@ title: Qualcomm Technologies Inc. LPASS CPU dai driver maintainers: - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> - - Rohit kumar <rohitkr(a)codeaurora.org> + - Rohit kumar <quic_rohkumar(a)quicinc.com> description: | Qualcomm Technologies Inc. SOC Low-Power Audio SubSystem (LPASS) that consist -- 2.25.1

2 1

[PATCH] ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode
by Shengjiu Wang 07 Jul '23

07 Jul '23

On i.MX8MM, the MCTL_MCLK_EN bit it is not only the gate for MCLK output to PAD, but also the gate bit between root clock and SAI module, So it is need to be enabled for master mode, otherwise there is no bclk generated. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fa950dde5310..dae16a14f177 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -437,6 +437,12 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP, savediv / 2 - 1); + if (sai->soc_data->max_register >= FSL_SAI_MCTL) { + /* SAI is in master mode at this point, so enable MCLK */ + regmap_update_bits(sai->regmap, FSL_SAI_MCTL, + FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); + } + return 0; } -- 2.17.1

5 11

[RFC] ASoC: fsl_sai: Do not set MCTL_MCLK_EN based on the number of registers
by Fabio Estevam 06 Jul '23

06 Jul '23

From: Fabio Estevam <festevam(a)denx.de> Since commit ff87d619ac18 ("ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode") Andreas reports that on an i.MX8MP-based system where MCLK needs to be an input, the MCLK pin is actually an output, despite not having the 'fsl,sai-mclk-direction-output' property present in the devicetree. This commit explains the following: "On i.MX8MM, the MCTL_MCLK_EN bit it is not only the gate for MCLK output to PAD, but also the gate bit between root clock and SAI module, So it is need to be enabled for master mode, otherwise there is no bclk generated." Currently, the decision of using MCTL_MCLK_EN as a clock gate is based on the number of SAI registers present on a platform. This is fragile as it causes issues on i.MX8MP, for example. Fix the problem by introducing a new boolean mclk_en_gates_clock member in the fsl_sai_soc_data structure that indicates that the MCTL_MCLK_EN bit also acts as a gate between the root clock and the SAI block. This allows i.MX8MM to turn on FSL_SAI_MCTL_MCLK_EN as intended and other SoCs such as i.MX8MP can still use MCLK as input. Fixes: ff87d619ac18 ("ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode") Reported-by: Andreas Henriksson <andreas(a)fatal.se> Signed-off-by: Fabio Estevam <festevam(a)denx.de> Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- Hi Shengjiu, Is imx8mm the only SoC that needs mclk_en_gates_clock = true? What about imx8mn and imx93? sound/soc/fsl/fsl_sai.c | 12 +++++++++++- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5e09f634c61b..e4dc6964f011 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -507,7 +507,7 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) savediv / 2 - 1); } - if (sai->soc_data->max_register >= FSL_SAI_MCTL) { + if (sai->soc_data->mclk_en_gates_clock) { /* SAI is in master mode at this point, so enable MCLK */ regmap_update_bits(sai->regmap, FSL_SAI_MCTL, FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); @@ -1513,6 +1513,7 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RMR, }; @@ -1524,6 +1525,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .reg_offset = 0, .mclk0_is_mclk1 = true, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RMR, }; @@ -1535,6 +1537,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .reg_offset = 8, .mclk0_is_mclk1 = false, .flags = PMQOS_CPU_LATENCY, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RMR, }; @@ -1546,6 +1549,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .reg_offset = 8, .mclk0_is_mclk1 = false, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RMR, }; @@ -1557,6 +1561,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RMR, }; @@ -1568,6 +1573,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mm_data = { .mclk0_is_mclk1 = false, .pins = 8, .flags = 0, + .mclk_en_gates_clock = true, .max_register = FSL_SAI_MCTL, }; @@ -1579,6 +1585,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mn_data = { .mclk0_is_mclk1 = false, .pins = 8, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_MDIV, }; @@ -1590,6 +1597,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mp_data = { .mclk0_is_mclk1 = false, .pins = 8, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_MDIV, .mclk_with_tere = true, }; @@ -1602,6 +1610,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8ulp_data = { .mclk0_is_mclk1 = false, .pins = 4, .flags = PMQOS_CPU_LATENCY, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RTCAP, }; @@ -1613,6 +1622,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx93_data = { .mclk0_is_mclk1 = false, .pins = 4, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_MCTL, .max_burst = {8, 8}, }; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 8254c3547b87..473260b0fbea 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -232,6 +232,7 @@ struct fsl_sai_soc_data { bool use_edma; bool mclk0_is_mclk1; bool mclk_with_tere; + bool mclk_en_gates_clock; unsigned int fifo_depth; unsigned int pins; unsigned int reg_offset; -- 2.34.1

1 1

[PATCH] ALSA: jack: Fix mutex call in snd_jack_report()
by Takashi Iwai 06 Jul '23

06 Jul '23

snd_jack_report() is supposed to be callable from an IRQ context, too, and it's indeed used in that way from virtsnd driver. The fix for input_dev race in commit 1b6a6fc5280e ("ALSA: jack: Access input_dev under mutex"), however, introduced a mutex lock in snd_jack_report(), and this resulted in a potential sleep-in-atomic. For addressing that problem, this patch changes the relevant code to use the object get/put and removes the mutex usage. That is, snd_jack_report(), it takes input_get_device() and leaves with input_put_device() for assuring the input_dev being assigned. Although the whole mutex could be reduced, we keep it because it can be still a protection for potential races between creation and deletion. Fixes: 1b6a6fc5280e ("ALSA: jack: Access input_dev under mutex") Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Closes: https://lore.kernel.org/r/cf95f7fe-a748-4990-8378-000491b40329@moroto.mount… Tested-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/jack.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/core/jack.c b/sound/core/jack.c index 88493cc31914..03d155ed362b 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -654,6 +654,7 @@ void snd_jack_report(struct snd_jack *jack, int status) struct snd_jack_kctl *jack_kctl; unsigned int mask_bits = 0; #ifdef CONFIG_SND_JACK_INPUT_DEV + struct input_dev *idev; int i; #endif @@ -670,17 +671,15 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - mutex_lock(&jack->input_dev_lock); - if (!jack->input_dev) { - mutex_unlock(&jack->input_dev_lock); + idev = input_get_device(jack->input_dev); + if (!idev) return; - } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits); if (jack->type & testbit) - input_report_key(jack->input_dev, jack->key[i], + input_report_key(idev, jack->key[i], status & testbit); } @@ -688,13 +687,13 @@ void snd_jack_report(struct snd_jack *jack, int status) int testbit = ((1 << i) & ~mask_bits); if (jack->type & testbit) - input_report_switch(jack->input_dev, + input_report_switch(idev, jack_switch_types[i], status & testbit); } - input_sync(jack->input_dev); - mutex_unlock(&jack->input_dev_lock); + input_sync(idev); + input_put_device(idev); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.3

1 0

[bug report] ALSA: jack: Access input_dev under mutex
by Dan Carpenter 06 Jul '23

06 Jul '23

Hello Amadeusz Sławiński, The patch 1b6a6fc5280e: "ALSA: jack: Access input_dev under mutex" from Apr 12, 2022, leads to the following Smatch static checker warning: sound/core/jack.c:673 snd_jack_report() warn: sleeping in atomic context sound/core/jack.c 663 jack->hw_status_cache = status; 664 665 list_for_each_entry(jack_kctl, &jack->kctl_list, list) 666 if (jack_kctl->sw_inject_enable) 667 mask_bits |= jack_kctl->mask_bits; 668 else 669 snd_kctl_jack_report(jack->card, jack_kctl->kctl, 670 status & jack_kctl->mask_bits); 671 672 #ifdef CONFIG_SND_JACK_INPUT_DEV --> 673 mutex_lock(&jack->input_dev_lock); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ That patch adds this mutex but we can't take mutex because we're already holding a spinlock. The problematic call trees are: virtsnd_event_notify_cb() <- disables preempt virtsnd_disable_event_vq() <- disables preempt -> virtsnd_event_dispatch() -> virtsnd_jack_event() -> snd_jack_report() The virtsnd_event_notify_cb() and virtsnd_disable_event_vq() functions take the spin_lock_irqsave(&queue->lock, flags); regards, dan carpenter

3 4

[PATCH] ASoC: qcom: q6afe-dai: fix Display Port Playback stream name
by Srinivas Kandagatla 06 Jul '23

06 Jul '23

With recent changes to add more display ports did not change the Stream name in q6afe-dai. This results in below error "ASoC: Failed to add route DISPLAY_PORT_RX -> Display Port Playback(*)" and sound card fails to probe. Fix this by adding correct stream name. Fixes: 90848a2557fe ("ASoC: qcom: q6dsp: add support to more display ports") Reported-by: Amit Pundir <amit.pundir(a)linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 31e0bad71e95..dbff55a97162 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -476,7 +476,7 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"HDMI Playback", NULL, "HDMI_RX"}, - {"Display Port Playback", NULL, "DISPLAY_PORT_RX"}, + {"DISPLAY_PORT_RX_0 Playback", NULL, "DISPLAY_PORT_RX"}, {"Slimbus Playback", NULL, "SLIMBUS_0_RX"}, {"Slimbus1 Playback", NULL, "SLIMBUS_1_RX"}, {"Slimbus2 Playback", NULL, "SLIMBUS_2_RX"}, -- 2.25.1

3 2

[PATCH v2 0/3] AMD Vangogh support for NAU8821/MAX98388
by Cristian Ciocaltea 06 Jul '23

06 Jul '23

This patch series extends the Vangogh machine driver to support a variant based on the Nuvoton NAU88L21 Codec and the Analog Devices MAX98388 Speaker Amplifier. Changes in v2: * Reworked series to ensure the existing devices based on the CS35L41 codec continue to work fine * Rebased onto next-20230703 Cristian Ciocaltea (3): ASoC: amd: vangogh: Make use of DRV_NAME ASoC: amd: vangogh: Use dmi_first_match() for DMI quirk handling ASoC: amd: vangogh: Add support for NAU8821/MAX98388 variant sound/soc/amd/Kconfig | 5 +- sound/soc/amd/vangogh/acp5x-mach.c | 167 +++++++++++++++++++++++++---- 2 files changed, 149 insertions(+), 23 deletions(-) -- 2.41.0

2 5

[tiwai-sound:topic/midi20-gadget 63/63] htmldocs: Documentation/usb/gadget-testing.rst:1019: WARNING: Malformed table.
by kernel test robot 05 Jul '23

05 Jul '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20-gadget head: be0c9761ecd79271d4a3efacaec6963a4010a297 commit: be0c9761ecd79271d4a3efacaec6963a4010a297 [63/63] usb: gadget: midi2: Add testing documentation reproduce: (https://download.01.org/0day-ci/archive/20230706/202307060422.HHYZIPqX-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307060422.HHYZIPqX-lkp@intel.com/ All warnings (new ones prefixed by >>): >> Documentation/usb/gadget-testing.rst:1019: WARNING: Malformed table. vim +1019 Documentation/usb/gadget-testing.rst 1003 1004 =============== =============================================== 1005 name Function Block name string 1006 direction Direction of this FB 1007 1: input, 2: output, or 3: bidirectional 1008 first_group The first UMP Group number (0-15) 1009 num_groups The number of groups in this FB (1-16) 1010 ui_hint UI-hint of this FB 1011 0: unknown, 1: receiver, 2: sender, 3: both 1012 midi_ci_verison Supported MIDI-CI version number (8 bit) 1013 is_midi1 Legacy MIDI 1.0 device (0-2) 1014 0: MIDI 2.0 device, 1015 1: MIDI 1.0 without restriction, or 1016 2: MIDI 1.0 with low speed 1017 sysex8_streams Max number of SysEx8 streams (8 bit) 1018 active Bool flag for FB activity (0 or 1) > 1019 ============= =============================================== 1020 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:test/snd_ctl_remove-lock-fix 40/54] htmldocs: ./include/sound/control.h:162: warning: expecting prototype for snd_ctl_find_id(). Prototype was for snd_ctl_find_id_mixer() instead
by kernel test robot 05 Jul '23

05 Jul '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/snd_ctl_remove-lock-fix head: 9ad18619a06d551105bd84191dca5bb5295277c7 commit: 59444a29dbfd44718fef15083aa339c2c8b579b5 [40/54] ALSA: control: Introduce snd_ctl_find_id_mixer() reproduce: (https://download.01.org/0day-ci/archive/20230706/202307060119.mTXEyy6N-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307060119.mTXEyy6N-lkp@intel.com/ All warnings (new ones prefixed by >>): >> ./include/sound/control.h:162: warning: expecting prototype for snd_ctl_find_id(). Prototype was for snd_ctl_find_id_mixer() instead vim +162 ./include/sound/control.h 147 148 /** 149 * snd_ctl_find_id - find the control instance with the given name string 150 * @card: the card instance 151 * @name: the name string 152 * 153 * Finds the control instance with the given name and 154 * @SNDRV_CTL_ELEM_IFACE_MIXER. Other fields are set to zero. 155 * 156 * This is merely a wrapper to snd_ctl_find_id(). 157 * 158 * Return: The pointer of the instance if found, or %NULL if not. 159 */ 160 static inline struct snd_kcontrol * 161 snd_ctl_find_id_mixer(struct snd_card *card, const char *name) > 162 { 163 struct snd_ctl_elem_id id = {}; 164 165 id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; 166 strscpy(id.name, name, sizeof(id.name)); 167 return snd_ctl_find_id(card, &id); 168 } 169 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v3 0/6] Introduce STM32 system bus
by Gatien Chevallier 05 Jul '23

05 Jul '23

Document STM32 System Bus. This bus is intended to control firewall access for the peripherals connected to it. For every peripheral, the bus checks the firewall registers to see if the peripheral is configured as non-secure. If the peripheral is configured as secure, the node is marked populated, so the device won't be probed. This is useful as a firewall configuration sanity check and avoid platform crashes in case peripherals are incorrectly configured. The STM32 System Bus implements the feature-domain-controller bindings. It is used by peripherals to reference a domain controller, in this case the firewall feature domain. The bus uses the ID referenced by the feature-domains property to know where to look in the firewall to get the security configuration for the peripheral. This allows a device tree description rather than a hardcoded peripheral table in the bus driver. On STM32MP13/15 platforms, the firewall bus is represented by the ETZPC node, which is responsible for the securing / MCU isolating the capable peripherals. STM32MP13/15 device trees are updated in this series to implement the bus. All peripherals that are securable or MCU isolation capable by the ETZPC are connected to the bus. Changes in V2: - Corrected YAMLS errors highlighted by Rob's robot - Re-ordered Signed-off-by tags in two patches Changes in V3: - Document feature-domains property in YAML documentation for concerned periperals under the System Bus - Fix STM32 System Bus YAML documentation - Remove STM32 System bus bindings that were currently used as helpers for device tree - Correct few errors in driver - Add missing peripherals under the System Bus that were in SoC variation device tree files - Fix node names Gatien Chevallier (5): dt-bindings: treewide: add feature-domains description in binding files dt-bindings: bus: add STM32 System Bus bus: stm32_sys_bus: add support for STM32MP15 and STM32MP13 system bus ARM: dts: stm32: add ETZPC as a system bus for STM32MP15x boards ARM: dts: stm32: add ETZPC as a system bus for STM32MP13x boards Oleksii Moisieiev (1): dt-bindings: Document common device controller bindings .../devicetree/bindings/bus/st,sys-bus.yaml | 127 + .../bindings/crypto/st,stm32-hash.yaml | 5 + .../devicetree/bindings/dma/st,stm32-dma.yaml | 5 + .../bindings/dma/st,stm32-dmamux.yaml | 5 + .../feature-domain-controller.yaml | 84 + .../devicetree/bindings/i2c/st,stm32-i2c.yaml | 5 + .../bindings/iio/adc/st,stm32-adc.yaml | 5 + .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 5 + .../bindings/iio/dac/st,stm32-dac.yaml | 5 + .../bindings/media/st,stm32-cec.yaml | 5 + .../bindings/media/st,stm32-dcmi.yaml | 5 + .../memory-controllers/st,stm32-fmc2-ebi.yaml | 5 + .../bindings/mfd/st,stm32-lptimer.yaml | 5 + .../bindings/mfd/st,stm32-timers.yaml | 6 + .../devicetree/bindings/mmc/arm,pl18x.yaml | 5 + .../devicetree/bindings/net/stm32-dwmac.yaml | 5 + .../bindings/phy/phy-stm32-usbphyc.yaml | 5 + .../bindings/regulator/st,stm32-vrefbuf.yaml | 5 + .../devicetree/bindings/rng/st,stm32-rng.yaml | 5 + .../bindings/serial/st,stm32-uart.yaml | 5 + .../bindings/sound/st,stm32-i2s.yaml | 5 + .../bindings/sound/st,stm32-sai.yaml | 5 + .../bindings/sound/st,stm32-spdifrx.yaml | 5 + .../bindings/spi/st,stm32-qspi.yaml | 5 + .../devicetree/bindings/spi/st,stm32-spi.yaml | 5 + .../devicetree/bindings/usb/dwc2.yaml | 5 + MAINTAINERS | 6 + arch/arm/boot/dts/stm32mp131.dtsi | 407 +-- arch/arm/boot/dts/stm32mp133.dtsi | 51 +- arch/arm/boot/dts/stm32mp13xc.dtsi | 19 +- arch/arm/boot/dts/stm32mp13xf.dtsi | 18 +- arch/arm/boot/dts/stm32mp151.dtsi | 2722 +++++++++-------- arch/arm/boot/dts/stm32mp153.dtsi | 52 +- arch/arm/boot/dts/stm32mp15xc.dtsi | 19 +- drivers/bus/Kconfig | 9 + drivers/bus/Makefile | 1 + drivers/bus/stm32_sys_bus.c | 168 + 37 files changed, 2208 insertions(+), 1596 deletions(-) create mode 100644 Documentation/devicetree/bindings/bus/st,sys-bus.yaml create mode 100644 Documentation/devicetree/bindings/feature-controllers/feature-domain-controller.yaml create mode 100644 drivers/bus/stm32_sys_bus.c -- 2.35.3

7 23

Garbage returned while reading unused S/PDIF, A CM106 like sound card
by Aurélien 05 Jul '23

05 Jul '23

Hello, I encounter troubles while using S/PDIF input on an CM106 like sound card (vendor 0x0d8c, device 0x0102) when the remote device which is connected by S/ PDIF to the sound card is powered off. In that case garbage data are reading from the S/PDIF rather than.. nothing It is really boring since this S/PDIF input is connected to the output speakers and once the remote deviced is powered off it emits a really disturbing sound. Since there is no input on the S/PDIF I guess that it could be a driver bug? If it's an hardware bug (I don't encounter the problem under windows..) is it possible to make a hack in order to distinguish between real data and noise on the S/PDIF input? Unfortunately this test is performed on an embedded device which runs linux 5.16.17-sun50iw6. I don't know if this problem still occurs with newer alsa version. Thank you in advance for your help. Regards, Aurélien

2 1

[PATCH 2/3] ASoC: rt5645: check return value after reading device id
by shumingf＠realtek.com 05 Jul '23

05 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> If the I2C controller encounters some problems like timed-out, the codec driver will report the error code for the first read. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index c0d56a8f7f47..b0953e9bcaf9 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3967,7 +3967,11 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) * read and power On. */ msleep(TIME_TO_POWER_MS); - regmap_read(regmap, RT5645_VENDOR_ID2, &val); + ret = regmap_read(regmap, RT5645_VENDOR_ID2, &val); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read: 0x%02X\n, ret = %d", RT5645_VENDOR_ID2, ret); + goto err_enable; + } switch (val) { case RT5645_DEVICE_ID: -- 2.41.0

2 1

[PATCH] ASoC: cs35l45: Select REGMAP_IRQ
by Nathan Chancellor 05 Jul '23

05 Jul '23

After commit 6085f9e6dc19 ("ASoC: cs35l45: IRQ support"), without any other configuration that selects CONFIG_REGMAP_IRQ, modpost errors out with: ERROR: modpost: "regmap_irq_get_virq" [sound/soc/codecs/snd-soc-cs35l45.ko] undefined! ERROR: modpost: "devm_regmap_add_irq_chip" [sound/soc/codecs/snd-soc-cs35l45.ko] undefined! Add the Kconfig selection to ensure these functions get built and included, which resolves the build failure. Cc: stable(a)vger.kernel.org Fixes: 6085f9e6dc19 ("ASoC: cs35l45: IRQ support") Reported-by: Marcus Seyfarth <m.seyfarth(a)gmail.com> Closes: https://github.com/ClangBuiltLinux/linux/issues/1882 Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0cd107fa112f..76ddd3ffc496 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -715,6 +715,7 @@ config SND_SOC_CS35L41_I2C config SND_SOC_CS35L45 tristate + select REGMAP_IRQ config SND_SOC_CS35L45_SPI tristate "Cirrus Logic CS35L45 CODEC (SPI)" --- base-commit: 6f49256897083848ce9a59651f6b53fc80462397 change-id: 20230703-cs35l45-select-regmap_irq-0fd9d631763c Best regards, -- Nathan Chancellor <nathan(a)kernel.org>

2 1

snd_soc_dai_ops.startup called for all codecs
by Peter Suti 05 Jul '23

05 Jul '23

Hi, I'm trying to create a sound card which has two codecs associated with one DAI. One codec is responsible for playback and the other one for capture. I can see in soc-pcm.c that __soc_pcm_hw_params() is checking if the stream is valid for the current codec with snd_soc_dai_stream_valid() and if it isn't it is not calling the hw_params() callback. But this check is not there in __soc_pcm_open() which calls snd_soc_dai_startup() for every codec. Can someone shed some light on this? Why is startup() called for codecs that don't support the current stream? Thank you! Best Regards, Peter Suti

3 3

[tiwai-sound:test/snd_ctl_remove-lock-fix 31/38] sound/soc/mediatek/mt8188/mt8188-mt6359.c:1000:25: error: implicit declaration of function 'snd_ctl_remove'; did you mean 'snd_ctl_rename'?
by kernel test robot 05 Jul '23

05 Jul '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/snd_ctl_remove-lock-fix head: 1daeb793bc0d3c3ad9a66718852a4be0584b700c commit: 664c9b8fd2f9e61cbe85d8015fc1962ed10d8a2a [31/38] ALSA: control: Drop snd_ctl_remove() config: arm-allmodconfig (https://download.01.org/0day-ci/archive/20230705/202307051751.ewoQ5wky-lkp@…) compiler: arm-linux-gnueabi-gcc (GCC) 12.3.0 reproduce: (https://download.01.org/0day-ci/archive/20230705/202307051751.ewoQ5wky-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307051751.ewoQ5wky-lkp@intel.com/ All errors (new ones prefixed by >>): sound/soc/mediatek/mt8188/mt8188-mt6359.c: In function 'mt8188_fixup_controls': >> sound/soc/mediatek/mt8188/mt8188-mt6359.c:1000:25: error: implicit declaration of function 'snd_ctl_remove'; did you mean 'snd_ctl_rename'? [-Werror=implicit-function-declaration] 1000 | snd_ctl_remove(card->snd_card, kctl); | ^~~~~~~~~~~~~~ | snd_ctl_rename cc1: some warnings being treated as errors vim +1000 sound/soc/mediatek/mt8188/mt8188-mt6359.c 9f08dcbddeb307 Trevor Wu 2023-05-26 981 9f08dcbddeb307 Trevor Wu 2023-05-26 982 static void mt8188_fixup_controls(struct snd_soc_card *card) 9f08dcbddeb307 Trevor Wu 2023-05-26 983 { 9f08dcbddeb307 Trevor Wu 2023-05-26 984 struct mt8188_mt6359_priv *priv = snd_soc_card_get_drvdata(card); 9f08dcbddeb307 Trevor Wu 2023-05-26 985 struct mt8188_card_data *card_data = (struct mt8188_card_data *)priv->private_data; 9f08dcbddeb307 Trevor Wu 2023-05-26 986 struct snd_kcontrol *kctl; 9f08dcbddeb307 Trevor Wu 2023-05-26 987 9f08dcbddeb307 Trevor Wu 2023-05-26 988 if (card_data->quirk & NAU8825_HS_PRESENT) { 9f08dcbddeb307 Trevor Wu 2023-05-26 989 struct snd_soc_dapm_widget *w, *next_w; 9f08dcbddeb307 Trevor Wu 2023-05-26 990 9f08dcbddeb307 Trevor Wu 2023-05-26 991 for_each_card_widgets_safe(card, w, next_w) { 9f08dcbddeb307 Trevor Wu 2023-05-26 992 if (strcmp(w->name, "Headphone")) 9f08dcbddeb307 Trevor Wu 2023-05-26 993 continue; 9f08dcbddeb307 Trevor Wu 2023-05-26 994 9f08dcbddeb307 Trevor Wu 2023-05-26 995 snd_soc_dapm_free_widget(w); 9f08dcbddeb307 Trevor Wu 2023-05-26 996 } 9f08dcbddeb307 Trevor Wu 2023-05-26 997 9f08dcbddeb307 Trevor Wu 2023-05-26 998 kctl = ctl_find(card->snd_card, "Headphone Switch"); 9f08dcbddeb307 Trevor Wu 2023-05-26 999 if (kctl) 9f08dcbddeb307 Trevor Wu 2023-05-26 @1000 snd_ctl_remove(card->snd_card, kctl); 9f08dcbddeb307 Trevor Wu 2023-05-26 1001 else 9f08dcbddeb307 Trevor Wu 2023-05-26 1002 dev_warn(card->dev, "Cannot find ctl : Headphone Switch\n"); 9f08dcbddeb307 Trevor Wu 2023-05-26 1003 } 9f08dcbddeb307 Trevor Wu 2023-05-26 1004 } 9f08dcbddeb307 Trevor Wu 2023-05-26 1005 :::::: The code at line 1000 was first introduced by commit :::::: 9f08dcbddeb307793bbfff036db213d2cdf03a50 ASoC: mediatek: mt8188-mt6359: support new board with nau88255 :::::: TO: Trevor Wu <trevor.wu(a)mediatek.com> :::::: CC: Mark Brown <broonie(a)kernel.org> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v1] sound: seq: check struct system assignment problem
by Minjie Du 05 Jul '23

05 Jul '23

make data->system.parm1 avoid double assignment. but note writes "RT message with two parameters to UMP" is one of assignment is for system.parm2? I'm not sure, Could you help check it out? Thank you! Signed-off-by: Minjie Du <duminjie(a)vivo.com> --- sound/core/seq/seq_ump_convert.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index eb1d86ff6..60e637149 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -713,7 +713,6 @@ static int system_2p_ev_to_ump_midi1(const struct snd_seq_event *event, unsigned char status) { data->system.status = status; - data->system.parm1 = (event->data.control.value >> 7) & 0x7f; data->system.parm1 = event->data.control.value & 0x7f; return 1; } -- 2.39.0

2 1

[tiwai-sound:test/snd_ctl_remove-lock-fix 34/38] drivers/staging/greybus/audio_codec.c:810:26: warning: variable 'card' set but not used
by kernel test robot 05 Jul '23

05 Jul '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/snd_ctl_remove-lock-fix head: 6dd4ae925ae5b90ccfbcb11dce06fd505cc18bc8 commit: 812cf704dac9f2f275d7bc53818fcccde62dcef8 [34/38] staging: greybus: Avoid abusing controls_rwsem config: x86_64-allyesconfig (https://download.01.org/0day-ci/archive/20230705/202307051355.BQPsIfxL-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce: (https://download.01.org/0day-ci/archive/20230705/202307051355.BQPsIfxL-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307051355.BQPsIfxL-lkp@intel.com/ All warnings (new ones prefixed by >>): drivers/staging/greybus/audio_codec.c: In function 'gbaudio_register_module': >> drivers/staging/greybus/audio_codec.c:810:26: warning: variable 'card' set but not used [-Wunused-but-set-variable] 810 | struct snd_card *card; | ^~~~ drivers/staging/greybus/audio_codec.c: In function 'gbaudio_unregister_module': >> drivers/staging/greybus/audio_codec.c:938:26: warning: unused variable 'card' [-Wunused-variable] 938 | struct snd_card *card = comp->card->snd_card; | ^~~~ vim +/card +810 drivers/staging/greybus/audio_codec.c 64a7e2cceb75cca drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 805 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 806 int gbaudio_register_module(struct gbaudio_module_info *module) d3d2af51f9c2f29 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 807 { 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 808 int ret; ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 809 struct snd_soc_component *comp; 2b8c2b51000e3d0 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-04-21 @810 struct snd_card *card; ac40b4d1e409971 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 811 struct gbaudio_jack *jack = NULL; d3d2af51f9c2f29 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 812 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 813 if (!gbcodec) { 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 814 dev_err(module->dev, "GB Codec not yet probed\n"); 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 815 return -EAGAIN; d3d2af51f9c2f29 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 816 } d3d2af51f9c2f29 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 817 ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 818 comp = gbcodec->component; ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 819 card = comp->card->snd_card; 2b8c2b51000e3d0 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-04-21 820 812cf704dac9f2f drivers/staging/greybus/audio_codec.c Takashi Iwai 2023-07-04 821 mutex_lock(&gbcodec->register_mutex); d3d2af51f9c2f29 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 822 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 823 if (module->num_dais) { 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 824 dev_err(gbcodec->dev, 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 825 "%d:DAIs not supported via gbcodec driver\n", 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 826 module->num_dais); 812cf704dac9f2f drivers/staging/greybus/audio_codec.c Takashi Iwai 2023-07-04 827 mutex_unlock(&gbcodec->register_mutex); 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 828 return -EINVAL; d3d2af51f9c2f29 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 829 } d3d2af51f9c2f29 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 830 ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 831 ret = gbaudio_init_jack(module, comp->card); 64a7e2cceb75cca drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 832 if (ret) { 812cf704dac9f2f drivers/staging/greybus/audio_codec.c Takashi Iwai 2023-07-04 833 mutex_unlock(&gbcodec->register_mutex); 64a7e2cceb75cca drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 834 return ret; 64a7e2cceb75cca drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 835 } 64a7e2cceb75cca drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 836 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 837 if (module->dapm_widgets) ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 838 snd_soc_dapm_new_controls(&comp->dapm, module->dapm_widgets, 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 839 module->num_dapm_widgets); 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 840 if (module->controls) ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 841 snd_soc_add_component_controls(comp, module->controls, 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 842 module->num_controls); 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 843 if (module->dapm_routes) ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 844 snd_soc_dapm_add_routes(&comp->dapm, module->dapm_routes, 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 845 module->num_dapm_routes); 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 846 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 847 /* card already instantiated, create widgets here only */ ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 848 if (comp->card->instantiated) { 620d28440c10b8f drivers/staging/greybus/audio_codec.c Deepak R Varma 2020-10-22 849 gbaudio_dapm_link_component_dai_widgets(comp->card, &comp->dapm); 64a7e2cceb75cca drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 850 #ifdef CONFIG_SND_JACK 461ab8077d573d4 drivers/staging/greybus/audio_codec.c Gioh Kim 2017-02-09 851 /* 461ab8077d573d4 drivers/staging/greybus/audio_codec.c Gioh Kim 2017-02-09 852 * register jack devices for this module 461ab8077d573d4 drivers/staging/greybus/audio_codec.c Gioh Kim 2017-02-09 853 * from codec->jack_list 461ab8077d573d4 drivers/staging/greybus/audio_codec.c Gioh Kim 2017-02-09 854 */ ac40b4d1e409971 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 855 list_for_each_entry(jack, &module->jack_list, list) { ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 856 snd_device_register(comp->card->snd_card, ac40b4d1e409971 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 857 jack->jack.jack); 64a7e2cceb75cca drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 858 } 64a7e2cceb75cca drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 859 #endif bb9986ef6b311fb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 860 } 64a7e2cceb75cca drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 861 c188fdc8176a8cb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 862 mutex_lock(&gbcodec->lock); 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 863 list_add(&module->list, &gbcodec->module_list); c188fdc8176a8cb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 864 mutex_unlock(&gbcodec->lock); c188fdc8176a8cb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 865 ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 866 if (comp->card->instantiated) ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 867 ret = snd_soc_dapm_new_widgets(comp->card); ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 868 dev_dbg(comp->dev, "Registered %s module\n", module->name); d3d2af51f9c2f29 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 869 812cf704dac9f2f drivers/staging/greybus/audio_codec.c Takashi Iwai 2023-07-04 870 mutex_unlock(&gbcodec->register_mutex); bb9986ef6b311fb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 871 return ret; d3d2af51f9c2f29 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 872 } 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 873 EXPORT_SYMBOL(gbaudio_register_module); d3d2af51f9c2f29 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 874 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 875 static void gbaudio_codec_clean_data_tx(struct gbaudio_data_connection *data) 25de3491f110648 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-01-13 876 { 79cb2b26f54e883 drivers/staging/greybus/audio_codec.c Chaehyun Lim 2016-09-20 877 u16 i2s_port, cportid; 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 878 int ret; d764212f73cf270 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 879 d764212f73cf270 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 880 if (list_is_singular(&gbcodec->module_list)) { 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 881 ret = gb_audio_apbridgea_stop_tx(data->connection, 0); 5bbe14b7acc2a00 drivers/staging/greybus/audio_codec.c Mark Greer 2016-02-29 882 if (ret) 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 883 return; 620d28440c10b8f drivers/staging/greybus/audio_codec.c Deepak R Varma 2020-10-22 884 ret = gb_audio_apbridgea_shutdown_tx(data->connection, 0); 25de3491f110648 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-01-13 885 if (ret) 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 886 return; d764212f73cf270 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 887 } 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 888 i2s_port = 0; /* fixed for now */ 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 889 cportid = data->connection->hd_cport_id; 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 890 ret = gb_audio_apbridgea_unregister_cport(data->connection, 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 891 i2s_port, cportid, 4a8e519902e73c8 drivers/staging/greybus/audio_codec.c Mark Greer 2016-02-26 892 AUDIO_APBRIDGEA_DIRECTION_TX); 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 893 data->state[0] = GBAUDIO_CODEC_SHUTDOWN; b19df7b9950cd8c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-02-16 894 } b19df7b9950cd8c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-02-16 895 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 896 static void gbaudio_codec_clean_data_rx(struct gbaudio_data_connection *data) 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 897 { 79cb2b26f54e883 drivers/staging/greybus/audio_codec.c Chaehyun Lim 2016-09-20 898 u16 i2s_port, cportid; 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 899 int ret; 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 900 d764212f73cf270 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 901 if (list_is_singular(&gbcodec->module_list)) { 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 902 ret = gb_audio_apbridgea_stop_rx(data->connection, 0); 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 903 if (ret) 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 904 return; 620d28440c10b8f drivers/staging/greybus/audio_codec.c Deepak R Varma 2020-10-22 905 ret = gb_audio_apbridgea_shutdown_rx(data->connection, 0); 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 906 if (ret) 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 907 return; d764212f73cf270 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 908 } 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 909 i2s_port = 0; /* fixed for now */ 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 910 cportid = data->connection->hd_cport_id; 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 911 ret = gb_audio_apbridgea_unregister_cport(data->connection, 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 912 i2s_port, cportid, 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 913 AUDIO_APBRIDGEA_DIRECTION_RX); 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 914 data->state[1] = GBAUDIO_CODEC_SHUTDOWN; 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 915 } 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 916 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 917 static void gbaudio_codec_cleanup(struct gbaudio_module_info *module) 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 918 { 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 919 struct gbaudio_data_connection *data; 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 920 int pb_state, cap_state; 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 921 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 922 dev_dbg(gbcodec->dev, "%s: removed, cleanup APBridge\n", module->name); 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 923 list_for_each_entry(data, &module->data_list, list) { 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 924 pb_state = data->state[0]; 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 925 cap_state = data->state[1]; 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 926 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 927 if (pb_state > GBAUDIO_CODEC_SHUTDOWN) 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 928 gbaudio_codec_clean_data_tx(data); 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 929 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 930 if (cap_state > GBAUDIO_CODEC_SHUTDOWN) 591c45227a77505 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-04 931 gbaudio_codec_clean_data_rx(data); 6339d2322c47f4b drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-01-13 932 } 6339d2322c47f4b drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-01-13 933 } 6339d2322c47f4b drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-01-13 934 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 935 void gbaudio_unregister_module(struct gbaudio_module_info *module) 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 936 { ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 937 struct snd_soc_component *comp = gbcodec->component; ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 @938 struct snd_card *card = comp->card->snd_card; ac40b4d1e409971 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 939 struct gbaudio_jack *jack, *n; 33cc283928249b2 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-08-05 940 int mask; b19df7b9950cd8c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-02-16 941 ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 942 dev_dbg(comp->dev, "Unregister %s module\n", module->name); b19df7b9950cd8c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-02-16 943 812cf704dac9f2f drivers/staging/greybus/audio_codec.c Takashi Iwai 2023-07-04 944 mutex_lock(&gbcodec->register_mutex); 6dd67645f22cfeb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 945 mutex_lock(&gbcodec->lock); bb9986ef6b311fb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 946 gbaudio_codec_cleanup(module); bb9986ef6b311fb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 947 list_del(&module->list); ce7bd8325dc75c8 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 948 dev_dbg(comp->dev, "Process Unregister %s module\n", module->name); c188fdc8176a8cb drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 949 mutex_unlock(&gbcodec->lock); 6339d2322c47f4b drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-01-13 950 :::::: The code at line 810 was first introduced by commit :::::: 2b8c2b51000e3d056f9aa27a64b93feabddf77a4 greybus: audio: Changes in response to ASoC cleanup :::::: TO: Vaibhav Agarwal <vaibhav.agarwal(a)linaro.org> :::::: CC: Greg Kroah-Hartman <gregkh(a)google.com> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[PATCH v2 0/5] ALSA: hda/realtek: add extra ROOG laptop quirks
by Luke D. Jones 05 Jul '23

05 Jul '23

- Adds quirks for various ASUS ROG laptops. - Fixes up a previous quirk - Fixes up a whitespace error This introduces a quirk chain that is similar to the ROG Ally quirk chain, but due to the verb configs being currently tested these new quirks do not use it and instead add a new chain. It is likely the Ally quirk will be removed in favour of this new chain in future. Luke D. Jones (5): ALSA: hda/realtek: Add quirk for ASUS ROG GX650P ALSA: hda/realtek: Add quirk for ASUS ROG GA402X ALSA: hda/realtek: Amend G634 quirk to enable rear speakers ALSA: hda/realtek: Add quirk for ASUS ROG G614Jx Fixes: 31278997add6 (ALSA: hda/realtek - Add headset quirk for Dell DT) sound/pci/hda/patch_realtek.c | 35 +++++++++++++++++++++++++++++++++-- 1 file changed, 33 insertions(+), 2 deletions(-) -- 2.41.0

3 7

[tiwai-sound:test/snd_ctl_remove-lock-fix 33/38] sound/soc/codecs/sigmadsp.c:676:45: error: no member named 'kctl' in 'struct sigmadsp_control'
by kernel test robot 04 Jul '23

04 Jul '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/snd_ctl_remove-lock-fix head: 6dd4ae925ae5b90ccfbcb11dce06fd505cc18bc8 commit: f6df559bc59a922fc6bd100cd018207323c3f9d1 [33/38] ASoC: sigmadsp: Simplify with snd_ctl_activate_id() config: hexagon-randconfig-r045-20230703 (https://download.01.org/0day-ci/archive/20230705/202307050346.R8ow8O6J-lkp@…) compiler: clang version 17.0.0 (https://github.com/llvm/llvm-project.git 4a5ac14ee968ff0ad5d2cc1ffa0299048db4c88a) reproduce: (https://download.01.org/0day-ci/archive/20230705/202307050346.R8ow8O6J-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307050346.R8ow8O6J-lkp@intel.com/ All errors (new ones prefixed by >>): In file included from sound/soc/codecs/sigmadsp.c:11: In file included from include/linux/i2c.h:19: In file included from include/linux/regulator/consumer.h:35: In file included from include/linux/suspend.h:5: In file included from include/linux/swap.h:9: In file included from include/linux/memcontrol.h:13: In file included from include/linux/cgroup.h:26: In file included from include/linux/kernel_stat.h:9: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:334: include/asm-generic/io.h:547:31: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 547 | val = __raw_readb(PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:560:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 560 | val = __le16_to_cpu((__le16 __force)__raw_readw(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:37:51: note: expanded from macro '__le16_to_cpu' 37 | #define __le16_to_cpu(x) ((__force __u16)(__le16)(x)) | ^ In file included from sound/soc/codecs/sigmadsp.c:11: In file included from include/linux/i2c.h:19: In file included from include/linux/regulator/consumer.h:35: In file included from include/linux/suspend.h:5: In file included from include/linux/swap.h:9: In file included from include/linux/memcontrol.h:13: In file included from include/linux/cgroup.h:26: In file included from include/linux/kernel_stat.h:9: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:334: include/asm-generic/io.h:573:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 573 | val = __le32_to_cpu((__le32 __force)__raw_readl(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:35:51: note: expanded from macro '__le32_to_cpu' 35 | #define __le32_to_cpu(x) ((__force __u32)(__le32)(x)) | ^ In file included from sound/soc/codecs/sigmadsp.c:11: In file included from include/linux/i2c.h:19: In file included from include/linux/regulator/consumer.h:35: In file included from include/linux/suspend.h:5: In file included from include/linux/swap.h:9: In file included from include/linux/memcontrol.h:13: In file included from include/linux/cgroup.h:26: In file included from include/linux/kernel_stat.h:9: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:334: include/asm-generic/io.h:584:33: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 584 | __raw_writeb(value, PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:594:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 594 | __raw_writew((u16 __force)cpu_to_le16(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:604:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 604 | __raw_writel((u32 __force)cpu_to_le32(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ >> sound/soc/codecs/sigmadsp.c:676:45: error: no member named 'kctl' in 'struct sigmadsp_control' 676 | changed = snd_ctl_activate_id(card, &ctrl->kctl->id, active); | ~~~~ ^ 6 warnings and 1 error generated. vim +676 sound/soc/codecs/sigmadsp.c 667 668 static void sigmadsp_activate_ctrl(struct sigmadsp *sigmadsp, 669 struct sigmadsp_control *ctrl, unsigned int samplerate_mask) 670 { 671 struct snd_card *card = sigmadsp->component->card->snd_card; 672 bool active; 673 int changed; 674 675 active = sigmadsp_samplerate_valid(ctrl->samplerates, samplerate_mask); > 676 changed = snd_ctl_activate_id(card, &ctrl->kctl->id, active); 677 if (active && changed > 0) { 678 mutex_lock(&sigmadsp->lock); 679 if (ctrl->cached) 680 sigmadsp_ctrl_write(sigmadsp, ctrl, ctrl->cache); 681 mutex_unlock(&sigmadsp->lock); 682 } 683 } 684 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:test/snd_ctl_remove-lock-fix 33/38] sound/soc/codecs/sigmadsp.c:676:50: error: 'struct sigmadsp_control' has no member named 'kctl'
by kernel test robot 04 Jul '23

04 Jul '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/snd_ctl_remove-lock-fix head: 6dd4ae925ae5b90ccfbcb11dce06fd505cc18bc8 commit: f6df559bc59a922fc6bd100cd018207323c3f9d1 [33/38] ASoC: sigmadsp: Simplify with snd_ctl_activate_id() config: nios2-randconfig-r026-20230703 (https://download.01.org/0day-ci/archive/20230705/202307050119.aWILsASi-lkp@…) compiler: nios2-linux-gcc (GCC) 12.3.0 reproduce: (https://download.01.org/0day-ci/archive/20230705/202307050119.aWILsASi-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307050119.aWILsASi-lkp@intel.com/ All errors (new ones prefixed by >>): sound/soc/codecs/sigmadsp.c: In function 'sigmadsp_activate_ctrl': >> sound/soc/codecs/sigmadsp.c:676:50: error: 'struct sigmadsp_control' has no member named 'kctl' 676 | changed = snd_ctl_activate_id(card, &ctrl->kctl->id, active); | ^~ vim +676 sound/soc/codecs/sigmadsp.c 667 668 static void sigmadsp_activate_ctrl(struct sigmadsp *sigmadsp, 669 struct sigmadsp_control *ctrl, unsigned int samplerate_mask) 670 { 671 struct snd_card *card = sigmadsp->component->card->snd_card; 672 bool active; 673 int changed; 674 675 active = sigmadsp_samplerate_valid(ctrl->samplerates, samplerate_mask); > 676 changed = snd_ctl_activate_id(card, &ctrl->kctl->id, active); 677 if (active && changed > 0) { 678 mutex_lock(&sigmadsp->lock); 679 if (ctrl->cached) 680 sigmadsp_ctrl_write(sigmadsp, ctrl, ctrl->cache); 681 mutex_unlock(&sigmadsp->lock); 682 } 683 } 684 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:test/snd_ctl_remove-lock-fix 34/38] drivers/staging/greybus/audio_codec.c:810:26: warning: variable 'card' set but not used
by kernel test robot 04 Jul '23

04 Jul '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/snd_ctl_remove-lock-fix head: 6dd4ae925ae5b90ccfbcb11dce06fd505cc18bc8 commit: 812cf704dac9f2f275d7bc53818fcccde62dcef8 [34/38] staging: greybus: Avoid abusing controls_rwsem config: alpha-allyesconfig (https://download.01.org/0day-ci/archive/20230705/202307050117.pUc9sZub-lkp@…) compiler: alpha-linux-gcc (GCC) 12.3.0 reproduce: (https://download.01.org/0day-ci/archive/20230705/202307050117.pUc9sZub-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307050117.pUc9sZub-lkp@intel.com/ All warnings (new ones prefixed by >>): drivers/staging/greybus/audio_codec.c: In function 'gbaudio_register_module': >> drivers/staging/greybus/audio_codec.c:810:26: warning: variable 'card' set but not used [-Wunused-but-set-variable] 810 | struct snd_card *card; | ^~~~ drivers/staging/greybus/audio_codec.c: In function 'gbaudio_unregister_module': drivers/staging/greybus/audio_codec.c:938:26: warning: unused variable 'card' [-Wunused-variable] 938 | struct snd_card *card = comp->card->snd_card; | ^~~~ vim +/card +810 drivers/staging/greybus/audio_codec.c 64a7e2cceb75cc drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 805 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 806 int gbaudio_register_module(struct gbaudio_module_info *module) d3d2af51f9c2f2 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 807 { 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 808 int ret; ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 809 struct snd_soc_component *comp; 2b8c2b51000e3d drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-04-21 @810 struct snd_card *card; ac40b4d1e40997 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 811 struct gbaudio_jack *jack = NULL; d3d2af51f9c2f2 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 812 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 813 if (!gbcodec) { 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 814 dev_err(module->dev, "GB Codec not yet probed\n"); 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 815 return -EAGAIN; d3d2af51f9c2f2 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 816 } d3d2af51f9c2f2 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 817 ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 818 comp = gbcodec->component; ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 819 card = comp->card->snd_card; 2b8c2b51000e3d drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-04-21 820 812cf704dac9f2 drivers/staging/greybus/audio_codec.c Takashi Iwai 2023-07-04 821 mutex_lock(&gbcodec->register_mutex); d3d2af51f9c2f2 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 822 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 823 if (module->num_dais) { 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 824 dev_err(gbcodec->dev, 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 825 "%d:DAIs not supported via gbcodec driver\n", 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 826 module->num_dais); 812cf704dac9f2 drivers/staging/greybus/audio_codec.c Takashi Iwai 2023-07-04 827 mutex_unlock(&gbcodec->register_mutex); 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 828 return -EINVAL; d3d2af51f9c2f2 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 829 } d3d2af51f9c2f2 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 830 ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 831 ret = gbaudio_init_jack(module, comp->card); 64a7e2cceb75cc drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 832 if (ret) { 812cf704dac9f2 drivers/staging/greybus/audio_codec.c Takashi Iwai 2023-07-04 833 mutex_unlock(&gbcodec->register_mutex); 64a7e2cceb75cc drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 834 return ret; 64a7e2cceb75cc drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 835 } 64a7e2cceb75cc drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 836 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 837 if (module->dapm_widgets) ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 838 snd_soc_dapm_new_controls(&comp->dapm, module->dapm_widgets, 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 839 module->num_dapm_widgets); 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 840 if (module->controls) ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 841 snd_soc_add_component_controls(comp, module->controls, 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 842 module->num_controls); 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 843 if (module->dapm_routes) ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 844 snd_soc_dapm_add_routes(&comp->dapm, module->dapm_routes, 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 845 module->num_dapm_routes); 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 846 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 847 /* card already instantiated, create widgets here only */ ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 848 if (comp->card->instantiated) { 620d28440c10b8 drivers/staging/greybus/audio_codec.c Deepak R Varma 2020-10-22 849 gbaudio_dapm_link_component_dai_widgets(comp->card, &comp->dapm); 64a7e2cceb75cc drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 850 #ifdef CONFIG_SND_JACK 461ab8077d573d drivers/staging/greybus/audio_codec.c Gioh Kim 2017-02-09 851 /* 461ab8077d573d drivers/staging/greybus/audio_codec.c Gioh Kim 2017-02-09 852 * register jack devices for this module 461ab8077d573d drivers/staging/greybus/audio_codec.c Gioh Kim 2017-02-09 853 * from codec->jack_list 461ab8077d573d drivers/staging/greybus/audio_codec.c Gioh Kim 2017-02-09 854 */ ac40b4d1e40997 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 855 list_for_each_entry(jack, &module->jack_list, list) { ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 856 snd_device_register(comp->card->snd_card, ac40b4d1e40997 drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 857 jack->jack.jack); 64a7e2cceb75cc drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 858 } 64a7e2cceb75cc drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 859 #endif bb9986ef6b311f drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 860 } 64a7e2cceb75cc drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 861 c188fdc8176a8c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 862 mutex_lock(&gbcodec->lock); 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 863 list_add(&module->list, &gbcodec->module_list); c188fdc8176a8c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 864 mutex_unlock(&gbcodec->lock); c188fdc8176a8c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 865 ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 866 if (comp->card->instantiated) ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 867 ret = snd_soc_dapm_new_widgets(comp->card); ce7bd8325dc75c drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2020-07-09 868 dev_dbg(comp->dev, "Registered %s module\n", module->name); d3d2af51f9c2f2 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 869 812cf704dac9f2 drivers/staging/greybus/audio_codec.c Takashi Iwai 2023-07-04 870 mutex_unlock(&gbcodec->register_mutex); bb9986ef6b311f drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-05-04 871 return ret; d3d2af51f9c2f2 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 872 } 6dd67645f22cfe drivers/staging/greybus/audio_codec.c Vaibhav Agarwal 2016-03-29 873 EXPORT_SYMBOL(gbaudio_register_module); d3d2af51f9c2f2 drivers/staging/greybus/audio-codec.c Vaibhav Agarwal 2015-11-23 874 :::::: The code at line 810 was first introduced by commit :::::: 2b8c2b51000e3d056f9aa27a64b93feabddf77a4 greybus: audio: Changes in response to ASoC cleanup :::::: TO: Vaibhav Agarwal <vaibhav.agarwal(a)linaro.org> :::::: CC: Greg Kroah-Hartman <gregkh(a)google.com> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] Add Dell XPS 15 9530 ALC289 speaker fix
by Pentracillin＠ 04 Jul '23

04 Jul '23

--- 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 afe8253f9..b5b80b6cd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9355,6 +9355,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0beb, "Dell XPS 15 9530", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), -- 2.34.1

2 1

[PATCH] ASoC: codecs: wcd938x: fix missing mbhc init error handling
by Johan Hovold 04 Jul '23

04 Jul '23

MBHC initialisation can fail so add the missing error handling to avoid dereferencing an error pointer when later configuring the jack: Unable to handle kernel paging request at virtual address fffffffffffffff8 pc : wcd_mbhc_start+0x28/0x380 [snd_soc_wcd_mbhc] lr : wcd938x_codec_set_jack+0x28/0x48 [snd_soc_wcd938x] Call trace: wcd_mbhc_start+0x28/0x380 [snd_soc_wcd_mbhc] wcd938x_codec_set_jack+0x28/0x48 [snd_soc_wcd938x] snd_soc_component_set_jack+0x28/0x8c [snd_soc_core] qcom_snd_wcd_jack_setup+0x7c/0x19c [snd_soc_qcom_common] sc8280xp_snd_init+0x20/0x2c [snd_soc_sc8280xp] snd_soc_link_init+0x28/0x90 [snd_soc_core] snd_soc_bind_card+0x628/0xbfc [snd_soc_core] snd_soc_register_card+0xec/0x104 [snd_soc_core] devm_snd_soc_register_card+0x4c/0xa4 [snd_soc_core] sc8280xp_platform_probe+0xf0/0x108 [snd_soc_sc8280xp] Fixes: bcee7ed09b8e ("ASoC: codecs: wcd938x: add Multi Button Headset Control support") Cc: stable(a)vger.kernel.org # 5.15 Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Johan Hovold <johan+linaro(a)kernel.org> --- sound/soc/codecs/wcd938x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 1595df2bf0e9..faa15a5ed2c8 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -2626,6 +2626,8 @@ static int wcd938x_mbhc_init(struct snd_soc_component *component) WCD938X_IRQ_HPHR_OCP_INT); wcd938x->wcd_mbhc = wcd_mbhc_init(component, &mbhc_cb, intr_ids, wcd_mbhc_fields, true); + if (IS_ERR(wcd938x->wcd_mbhc)) + return PTR_ERR(wcd938x->wcd_mbhc); snd_soc_add_component_controls(component, impedance_detect_controls, ARRAY_SIZE(impedance_detect_controls)); -- 2.39.3

3 2

[PATCH v1] sound: riptide: Remove duplicate judgments code
by Minjie Du 04 Jul '23

04 Jul '23

Fix: delate duplicate judgments Could you help check it out? Thank you! Signed-off-by: Minjie Du <duminjie(a)vivo.com> --- sound/pci/riptide/riptide.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b37c877c2..9bf4b2e86 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -958,8 +958,7 @@ getsourcesink(struct cmdif *cif, unsigned char source, unsigned char sink, { union cmdret rptr = CMDRET_ZERO; - if (SEND_RSSV(cif, source, sink, &rptr) && - SEND_RSSV(cif, source, sink, &rptr)) + if (SEND_RSSV(cif, source, sink, &rptr)) return -EIO; *a = rptr.retbytes[0]; *b = rptr.retbytes[1]; @@ -978,8 +977,7 @@ getsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int *rate) s = intdec; for (i = 0; i < 2; i++) { if (*s != 0xff) { - if (SEND_RSRC(cif, *s, &rptr) && - SEND_RSRC(cif, *s, &rptr)) + if (SEND_RSRC(cif, *s, &rptr)) return -EIO; p[i] += rptr.retwords[1]; p[i] *= rptr.retwords[2]; @@ -1013,8 +1011,7 @@ setsampleformat(struct cmdif *cif, sig = snd_pcm_format_unsigned(format) != 0; order = snd_pcm_format_big_endian(format) != 0; - if (SEND_SETF(cif, mixer, w, ch, order, sig, id) && - SEND_SETF(cif, mixer, w, ch, order, sig, id)) { + if (SEND_SETF(cif, mixer, w, ch, order, sig, id)) { snd_printdd("setsampleformat failed\n"); return -EIO; } @@ -1060,7 +1057,7 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval, { union cmdret rptr = CMDRET_ZERO; - if (SEND_RDGV(cif, num, num, &rptr) && SEND_RDGV(cif, num, num, &rptr)) + if (SEND_RDGV(cif, num, num, &rptr)) return -EIO; *rval = rptr.retwords[0]; *lval = rptr.retwords[1]; -- 2.39.0

3 2

[PATCH 1/5] ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 04 Jul '23

04 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> During ClockStop Mode0, peripheral interrupts are disabled. When system level resume is invoked, Peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 67404f45389f..4968a8c0064d 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -750,8 +750,15 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) if (!rt5682->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt5682->disable_irq == true) { + mutex_lock(&rt5682->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt5682->disable_irq = false; + mutex_unlock(&rt5682->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); -- 2.40.1

4 16

[PATCH v2 0/7] ASoC: mediatek: Add support for MT7986 SoC
by Maso Huang 04 Jul '23

04 Jul '23

Changes in v2: - v1 title: [PATCH 0/7] ASoC: mediatek: Add support for MT79xx SoC - add missing maintainers - rename mt79xx to mt7986 in all files - use clk bulk api in mt7986-afe-clk.c [2/7] - refine mt7986-afe-pcm.c based on reviewer's suggestions [4/7] - refine dt-binding files based on reviewer's suggestions [6/7] [7/7] - transpose [3/7] and [4/7] in v1 to fix test build errors This series of patches adds support for MediaTek AFE of MT7986 SoC. Patches are based on broonie tree "for-next" branch. Maso Huang (7): ASoC: mediatek: mt7986: add common header ASoC: mediatek: mt7986: support audio clock control ASoC: mediatek: mt7986: support etdm in platform driver ASoC: mediatek: mt7986: add platform driver ASoC: mediatek: mt7986: add machine driver with wm8960 ASoC: dt-bindings: mediatek,mt7986-wm8960: add mt7986-wm8960 document ASoC: dt-bindings: mediatek,mt7986-afe: add audio afe document .../bindings/sound/mediatek,mt7986-afe.yaml | 89 +++ .../sound/mediatek,mt7986-wm8960.yaml | 53 ++ sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt7986/Makefile | 10 + sound/soc/mediatek/mt7986/mt7986-afe-clk.c | 75 +++ sound/soc/mediatek/mt7986/mt7986-afe-clk.h | 18 + sound/soc/mediatek/mt7986/mt7986-afe-common.h | 51 ++ sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 598 ++++++++++++++++++ sound/soc/mediatek/mt7986/mt7986-dai-etdm.c | 421 ++++++++++++ sound/soc/mediatek/mt7986/mt7986-reg.h | 206 ++++++ sound/soc/mediatek/mt7986/mt7986-wm8960.c | 184 ++++++ 12 files changed, 1726 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml create mode 100644 sound/soc/mediatek/mt7986/Makefile create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-clk.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-clk.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-common.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-pcm.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-dai-etdm.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-reg.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-wm8960.c -- 2.18.0

4 15

[PATCH] ALSA: hda: fix a possible null-pointer dereference due to data race in snd_hdac_regmap_sync()
by Tuo Li 04 Jul '23

04 Jul '23

The variable codec->regmap is often protected by the lock codec->regmap_lock when is accessed. However, it is accessed without holding the lock when is accessed in snd_hdac_regmap_sync(): if (codec->regmap) In my opinion, this may be a harmful race, because if codec->regmap is set to NULL right after the condition is checked, a null-pointer dereference can occur in the called function regcache_sync(): map->lock(map->lock_arg); --> Line 360 in drivers/base/regmap/regcache.c To fix this possible null-pointer dereference caused by data race, the mutex_lock coverage is extended to protect the if statement as well as the function call to regcache_sync(). Reported-by: BassCheck <bass(a)buaa.edu.cn> Signed-off-by: Tuo Li <islituo(a)gmail.com> --- sound/hda/hdac_regmap.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index fe3587547cfe..39610a15bcc9 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -597,10 +597,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw_once); */ void snd_hdac_regmap_sync(struct hdac_device *codec) { - if (codec->regmap) { - mutex_lock(&codec->regmap_lock); + mutex_lock(&codec->regmap_lock); + if (codec->regmap) regcache_sync(codec->regmap); - mutex_unlock(&codec->regmap_lock); - } + mutex_unlock(&codec->regmap_lock); } EXPORT_SYMBOL_GPL(snd_hdac_regmap_sync); -- 2.34.1

2 2

[PATCH] ASoC: mediatek: mt8188: add constraints for PCM
by Trevor Wu 04 Jul '23

04 Jul '23

For certain projects, only a limited number of parameters are necessary. Therefore, it may be beneficial to add constraints that restrict the capacity of the PCM. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 145 +++++++++++++++++++++- 1 file changed, 144 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index ac69c23e0da1..0e4e1b8c99e2 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -30,6 +30,7 @@ #define TEST_MISO_DONE_2 BIT(29) #define NAU8825_HS_PRESENT BIT(0) +#define DPCM_FE_CONSTRAINT BIT(16) /* * Maxim MAX98390 @@ -452,6 +453,143 @@ enum { DAI_LINK_UL_SRC_BE, }; +static int mt8188_playback_startup(struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 48000 + }; + static const unsigned int channels[] = { + 2 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct mt8188_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mt8188_card_data *data = (struct mt8188_card_data *)priv->private_data; + int ret; + + if (!(data->quirk & DPCM_FE_CONSTRAINT)) + return 0; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list rate failed\n"); + return ret; + } + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list channel failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops mt8188_playback_ops = { + .startup = mt8188_playback_startup, +}; + +static int mt8188_capture_startup(struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 48000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct mt8188_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mt8188_card_data *data = (struct mt8188_card_data *)priv->private_data; + int ret; + + if (!(data->quirk & DPCM_FE_CONSTRAINT)) + return 0; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list rate failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops mt8188_capture_ops = { + .startup = mt8188_capture_startup, +}; + +static int mt8188_hdmi_dptx_startup(struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 48000 + }; + static const unsigned int channels[] = { + 2, 4, 6, 8 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct mt8188_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mt8188_card_data *data = (struct mt8188_card_data *)priv->private_data; + int ret; + + if (!(data->quirk & DPCM_FE_CONSTRAINT)) + return 0; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list rate failed\n"); + return ret; + } + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list channel failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops mt8188_hdmi_dptx_playback_ops = { + .startup = mt8188_hdmi_dptx_startup, +}; + static int mt8188_dptx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -723,6 +861,7 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8188_playback_ops, SND_SOC_DAILINK_REG(playback2), }, [DAI_LINK_DL3_FE] = { @@ -734,6 +873,7 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8188_playback_ops, SND_SOC_DAILINK_REG(playback3), }, [DAI_LINK_DL6_FE] = { @@ -778,6 +918,7 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8188_hdmi_dptx_playback_ops, SND_SOC_DAILINK_REG(playback10), }, [DAI_LINK_DL11_FE] = { @@ -833,6 +974,7 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8188_capture_ops, SND_SOC_DAILINK_REG(capture4), }, [DAI_LINK_UL5_FE] = { @@ -844,6 +986,7 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8188_capture_ops, SND_SOC_DAILINK_REG(capture5), }, [DAI_LINK_UL6_FE] = { @@ -1122,7 +1265,7 @@ static struct mt8188_card_data mt8188_evb_card = { static struct mt8188_card_data mt8188_nau8825_card = { .name = "mt8188_nau8825", - .quirk = NAU8825_HS_PRESENT, + .quirk = NAU8825_HS_PRESENT | DPCM_FE_CONSTRAINT, }; static const struct of_device_id mt8188_mt6359_dt_match[] = { -- 2.18.0

4 5

[PATCH v1 1/3] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 03 Jul '23

03 Jul '23

Integrate tas2781 configs for Lenovo Laptops. All of the tas2781s in the laptop will be aggregated as one speaker. The code support realtek as the primary codec. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v1: - remove white space at the end of the line in alc_fixup_headset_mode_alc255_no_hp_mic - Add tiwai(a)suse.de into Cc list - remove useless index - combine ALC287_FIXUP_TAS2781_I2C_2 and ALC287_FIXUP_TAS2781_I2C_4 together as ALC287_FIXUP_TAS2781_I2C, The code view all the tas2781s in the laptop as one instance. - delete the white space at the end of the line in alc_fixup_headset_mode_alc255_no_hp_mic - fix Laptop 0x17aa38be get the wrong fixup_id and enter into the wrong entry. --- sound/pci/hda/patch_realtek.c | 114 ++++++++++++++++++++++++++++++++-- 1 file changed, 110 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index afe8253f9a4f..2e5354a5decf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5883,7 +5883,7 @@ static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, struct alc_spec *spec = codec->spec; spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; alc255_set_default_jack_type(codec); - } + } else alc_fixup_headset_mode(codec, fix, action); } @@ -6705,7 +6705,7 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ } } -struct cs35l41_dev_name { +struct scodec_dev_name { const char *bus; const char *hid; int index; @@ -6714,7 +6714,7 @@ struct cs35l41_dev_name { /* match the device name in a slightly relaxed manner */ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) { - struct cs35l41_dev_name *p = data; + struct scodec_dev_name *p = data; const char *d = dev_name(dev); int n = strlen(p->bus); char tmp[32]; @@ -6730,12 +6730,32 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +static int comp_match_tas2781_dev_name(struct device *dev, + void *data) +{ + struct scodec_dev_name *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->bus, n)) + return 0; + /* skip the bus number */ + if (isdigit(d[n])) + n++; + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); + + return !strcmp(d + n, tmp); +} + static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct cs35l41_dev_name *rec; + struct scodec_dev_name *rec; int ret, i; switch (action) { @@ -6763,6 +6783,41 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } } +static void tas2781_generic_fixup(struct hda_codec *cdc, int action, + const char *bus, const char *hid) +{ + struct device *dev = hda_codec_dev(cdc); + struct alc_spec *spec = cdc->spec; + struct scodec_dev_name *rec; + int ret; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); + if (!rec) + return; + rec->bus = bus; + rec->hid = hid; + rec->index = 0; + spec->comps[0].codec = cdc; + component_match_add(dev, &spec->match, + comp_match_tas2781_dev_name, rec); + ret = component_master_add_with_match(dev, &comp_master_ops, + spec->match); + if (ret) + codec_err(cdc, + "Fail to register component aggregator %d\n", + ret); + else + spec->gen.pcm_playback_hook = + comp_generic_playback_hook; + break; + case HDA_FIXUP_ACT_FREE: + component_master_del(dev, &comp_master_ops); + break; + } +} + static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); @@ -6790,6 +6845,12 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); } +static void tas2781_fixup_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7210,6 +7271,7 @@ enum { ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, + ALC287_FIXUP_TAS2781_I2C, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9255,6 +9317,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC287_FIXUP_TAS2781_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9813,6 +9881,33 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", + ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -10728,6 +10823,17 @@ static int patch_alc269(struct hda_codec *codec) codec->fixup_id = HDA_FIXUP_ID_NOT_SET; } + /* FIXME: Laptop 0x17aa38be will get the wrong fixup_id and + * enter into the wrong entry. + * Correct the wrong entry. + */ + if (codec->fixup_id == ALC287_FIXUP_YOGA7_14ITL_SPEAKERS && + codec->core.vendor_id == 0x10ec0287 && + codec->core.subsystem_id == 0x17aa38be) { + codec_dbg(codec, "Clear wrong fixup for 17aa38be\n"); + codec->fixup_id = ALC287_FIXUP_TAS2781_I2C; + } + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true); snd_hda_pick_pin_fixup(codec, alc269_fallback_pin_fixup_tbl, alc269_fixups, false); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, -- 2.34.1

2 4

[PATCH] ASoC: codecs: wcd938x: fix soundwire initialisation race
by Johan Hovold 03 Jul '23

03 Jul '23

Make sure that the soundwire device used for register accesses has been enumerated and initialised before trying to read the codec variant during component probe. This specifically avoids interpreting (a masked and shifted) -EBUSY errno as the variant: wcd938x_codec audio-codec: ASoC: error at soc_component_read_no_lock on audio-codec for register: [0x000034b0] -16 in case the soundwire device has not yet been initialised, which in turn prevents some headphone controls from being registered. Fixes: 8d78602aa87a ("ASoC: codecs: wcd938x: add basic driver") Cc: stable(a)vger.kernel.org # 5.14 Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reported-by: Steev Klimaszewski <steev(a)kali.org> Signed-off-by: Johan Hovold <johan+linaro(a)kernel.org> --- sound/soc/codecs/wcd938x.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index e3ae4fb2c4db..4571588fad62 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3080,9 +3080,18 @@ static int wcd938x_irq_init(struct wcd938x_priv *wcd, struct device *dev) static int wcd938x_soc_codec_probe(struct snd_soc_component *component) { struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + struct sdw_slave *tx_sdw_dev = wcd938x->tx_sdw_dev; struct device *dev = component->dev; + unsigned long time_left; int ret, i; + time_left = wait_for_completion_timeout(&tx_sdw_dev->initialization_complete, + msecs_to_jiffies(2000)); + if (!time_left) { + dev_err(dev, "soundwire device init timeout\n"); + return -ETIMEDOUT; + } + snd_soc_component_init_regmap(component, wcd938x->regmap); ret = pm_runtime_resume_and_get(dev); -- 2.39.3

3 2

[PATCH 0/5] ASoC: codecs: wcd938x/wcd934x: loglevel fix and cleanups
by Johan Hovold 03 Jul '23

03 Jul '23

When investigating a race in the wcd938x driver I noticed that the MBHC impedance measurements where printed at error loglevel which is clearly wrong. Fix that, and clean up the logging somewhat by using dev_printk() and addressing some style issues. Included are also two patches that drop the bogus inline keywords from the functions involved. Johan Johan Hovold (5): ASoC: codecs: wcd938x: fix mbhc impedance loglevel ASoC: codecs: wcd938x: drop inline keywords ASoC: codecs: wcd938x: use dev_printk() for impedance logging ASoC: codecs: wcd934x: demote impedance printk loglevel ASoC: codecs: wcd934x: drop inline keywords sound/soc/codecs/wcd934x.c | 8 ++++---- sound/soc/codecs/wcd938x.c | 17 +++++++++-------- 2 files changed, 13 insertions(+), 12 deletions(-) -- 2.39.3

3 9

[PATCH] ASoC: qdsp6: q6apm: use dai link pcm id as pcm device number
by Srinivas Kandagatla 03 Jul '23

03 Jul '23

For some reason we ended up with a setup without this flag. This resulted in inconsistent sound card devices numbers which are also not starting as expected at dai_link->id. (Ex: MultiMedia1 pcm ended up with device number 4 instead of 0) With this patch patch now the MultiMedia1 PCM ends up with device number 0 as expected. Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support") Cc: <Stable(a)vger.kernel.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm-dai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 5eb0b864c740..c90db6daabbd 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -840,6 +840,7 @@ static const struct snd_soc_component_driver q6apm_fe_dai_component = { .pointer = q6apm_dai_pointer, .trigger = q6apm_dai_trigger, .compress_ops = &q6apm_dai_compress_ops, + .use_dai_pcm_id = true, }; static int q6apm_dai_probe(struct platform_device *pdev) -- 2.21.0

5 17

[PATCH] ALSA: pcm: Fix potential data race at PCM memory allocation helpers
by Takashi Iwai 03 Jul '23

03 Jul '23

The PCM memory allocation helpers have a sanity check against too many buffer allocations. However, the check is performed without a proper lock and the allocation isn't serialized; this allows user to allocate more memories than predefined max size. Practically seen, this isn't really a big problem, as it's more or less some "soft limit" as a sanity check, and it's not possible to allocate unlimitedly. But it's still better to address this for more consistent behavior. The patch covers the size check in do_alloc_pages() with the card->memory_mutex, and increases the allocated size there for preventing the further overflow. When the actual allocation fails, the size is decreased accordingly. Reported-by: BassCheck <bass(a)buaa.edu.cn> Reported-by: Tuo Li <islituo(a)gmail.com> Link: https://lore.kernel.org/r/CADm8Tek6t0WedK+3Y6rbE5YEt19tML8BUL45N2ji4ZAz1KcN… Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_memory.c | 44 +++++++++++++++++++++++++++++++++-------- 1 file changed, 36 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 7bde7fb64011..a0b951471699 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -31,15 +31,41 @@ static unsigned long max_alloc_per_card = 32UL * 1024UL * 1024UL; module_param(max_alloc_per_card, ulong, 0644); MODULE_PARM_DESC(max_alloc_per_card, "Max total allocation bytes per card."); +static void __update_allocated_size(struct snd_card *card, ssize_t bytes) +{ + card->total_pcm_alloc_bytes += bytes; +} + +static void update_allocated_size(struct snd_card *card, ssize_t bytes) +{ + mutex_lock(&card->memory_mutex); + __update_allocated_size(card, bytes); + mutex_unlock(&card->memory_mutex); +} + +static void decrease_allocated_size(struct snd_card *card, size_t bytes) +{ + mutex_lock(&card->memory_mutex); + WARN_ON(card->total_pcm_alloc_bytes < bytes); + __update_allocated_size(card, -(ssize_t)bytes); + mutex_unlock(&card->memory_mutex); +} + static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, int str, size_t size, struct snd_dma_buffer *dmab) { enum dma_data_direction dir; int err; + /* check and reserve the requested size */ + mutex_lock(&card->memory_mutex); if (max_alloc_per_card && - card->total_pcm_alloc_bytes + size > max_alloc_per_card) + card->total_pcm_alloc_bytes + size > max_alloc_per_card) { + mutex_unlock(&card->memory_mutex); return -ENOMEM; + } + __update_allocated_size(card, size); + mutex_unlock(&card->memory_mutex); if (str == SNDRV_PCM_STREAM_PLAYBACK) dir = DMA_TO_DEVICE; @@ -47,9 +73,14 @@ static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, dir = DMA_FROM_DEVICE; err = snd_dma_alloc_dir_pages(type, dev, dir, size, dmab); if (!err) { - mutex_lock(&card->memory_mutex); - card->total_pcm_alloc_bytes += dmab->bytes; - mutex_unlock(&card->memory_mutex); + /* the actual allocation size might be bigger than requested, + * and we need to correct the account + */ + if (dmab->bytes != size) + update_allocated_size(card, dmab->bytes - size); + } else { + /* take back on allocation failure */ + decrease_allocated_size(card, size); } return err; } @@ -58,10 +89,7 @@ static void do_free_pages(struct snd_card *card, struct snd_dma_buffer *dmab) { if (!dmab->area) return; - mutex_lock(&card->memory_mutex); - WARN_ON(card->total_pcm_alloc_bytes < dmab->bytes); - card->total_pcm_alloc_bytes -= dmab->bytes; - mutex_unlock(&card->memory_mutex); + decrease_allocated_size(card, dmab->bytes); snd_dma_free_pages(dmab); dmab->area = NULL; } -- 2.35.3

1 0

[PATCH 5/5] ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 03 Jul '23

03 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> During ClockStop Mode0, peripheral interrupts are disabled. When system level resume is invoked, Peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt722-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index cc57e4e27805..e9103ffb3f50 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -463,8 +463,16 @@ static int __maybe_unused rt722_sdca_dev_resume(struct device *dev) if (!rt722->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt722->disable_irq == true) { + mutex_lock(&rt722->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt722->disable_irq = false; + mutex_unlock(&rt722->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT722_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH 4/5] ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 03 Jul '23

03 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> During ClockStop Mode0, peripheral interrupts are disabled. When system level resume is invoked, Peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt712-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index ad06267b0ea0..6bc50396a0f6 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -438,8 +438,16 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev) if (!rt712->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt712->disable_irq == true) { + mutex_lock(&rt712->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt712->disable_irq = false; + mutex_unlock(&rt712->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH 3/5] ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 03 Jul '23

03 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> During ClockStop Mode0, peripheral interrupts are disabled. When system level resume is invoked, Peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 119e1f9605d7..23f23f714b39 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -438,8 +438,16 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH 2/5] ASoC: rt711: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 03 Jul '23

03 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> During ClockStop Mode0, peripheral interrupts are disabled. When system level resume is invoked, Peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 87dafcb4545d..33dced388f9e 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -538,8 +538,15 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH 0/5] fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 03 Jul '23

03 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> Currently, the codec driver could re-enable SDCA interrupt mask when the devices are re-attached. If the manager uses ClockStop Mode0 to peripherals, the SDCA interrupt mask will not be enabled when the system resumes. These patches will fix this issue. Shuming Fan (5): ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0 ASoC: rt711: fix for JD event handling in ClockStop Mode0 ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0 ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0 ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0 sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- sound/soc/codecs/rt712-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/rt722-sdca-sdw.c | 10 +++++++++- 5 files changed, 43 insertions(+), 5 deletions(-) -- 2.40.1

1 0

[PATCH v1 0/4] ASoC: remove copy of intlog10()
by Andy Shevchenko 03 Jul '23

03 Jul '23

The first three patches moves intlog10() to be available in entire kernel. The last one removes copy of it in one driver. Besides already good Lines of Code (LoC) statistics the upcoming users, if any, can utilize the exported functions. The series can be routed either via ASoC tree or media tree. Note, int_log.h is separated from math.h due to licensing. I dunno if we can mix two in a single header file. In any case we may do it later on. Andy Shevchenko (4): lib/math: Move dvb_math.c into lib/math/int_log.c lib/math/int_log: Use ARRAY_SIZE(logtable) where makes sense lib/math/int_log: Replace LGPL-2.1-or-later boilerplate with SPDX identifier ASoC: nau8825: Replace copied'n'pasted intlog10() Documentation/driver-api/media/dtv-common.rst | 2 +- drivers/media/dvb-core/Makefile | 2 +- drivers/media/dvb-frontends/af9013_priv.h | 2 +- drivers/media/dvb-frontends/af9033_priv.h | 2 +- drivers/media/dvb-frontends/cxd2820r_priv.h | 2 +- drivers/media/dvb-frontends/cxd2841er.c | 2 +- .../cxd2880/cxd2880_tnrdmd_dvbt2_mon.c | 2 +- .../cxd2880/cxd2880_tnrdmd_dvbt_mon.c | 2 +- .../media/dvb-frontends/cxd2880/cxd2880_top.c | 2 +- drivers/media/dvb-frontends/dib7000p.c | 2 +- drivers/media/dvb-frontends/dib8000.c | 2 +- drivers/media/dvb-frontends/dib9000.c | 2 +- drivers/media/dvb-frontends/drxk_hard.c | 2 +- drivers/media/dvb-frontends/lgdt3305.c | 2 +- drivers/media/dvb-frontends/lgdt3306a.c | 2 +- drivers/media/dvb-frontends/lgdt330x.c | 2 +- drivers/media/dvb-frontends/m88ds3103_priv.h | 2 +- drivers/media/dvb-frontends/mn88443x.c | 2 +- drivers/media/dvb-frontends/mn88472_priv.h | 2 +- drivers/media/dvb-frontends/mn88473_priv.h | 2 +- drivers/media/dvb-frontends/or51132.c | 2 +- drivers/media/dvb-frontends/or51211.c | 2 +- drivers/media/dvb-frontends/rtl2830_priv.h | 2 +- drivers/media/dvb-frontends/rtl2832_priv.h | 2 +- drivers/media/dvb-frontends/si2165.c | 2 +- drivers/media/dvb-frontends/stv0367.c | 2 +- drivers/media/dvb-frontends/tc90522.c | 2 +- drivers/media/dvb-frontends/tda10048.c | 2 +- include/{media/dvb_math.h => linux/int_log.h} | 18 +--- lib/math/Makefile | 2 +- .../dvb-core/dvb_math.c => lib/math/int_log.c | 26 ++---- sound/soc/codecs/nau8825.c | 93 +------------------ 32 files changed, 45 insertions(+), 150 deletions(-) rename include/{media/dvb_math.h => linux/int_log.h} (63%) rename drivers/media/dvb-core/dvb_math.c => lib/math/int_log.c (84%) -- 2.40.0.1.gaa8946217a0b

3 9

ucm: mark internal functions static
by GitHub pull_request - edited 01 Jul '23

01 Jul '23

alsa-project/alsa-lib pull request #335 was edited from neobrain: I've ran into build issues in a project that links against both libalsa and libxkbcommon, which both export a function called `parse_string`. Changing internal functions to internal linkage (`static`) prevents this kind of problem. Request URL : https://github.com/alsa-project/alsa-lib/pull/335 Patch URL : https://github.com/alsa-project/alsa-lib/pull/335.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ucm: mark internal functions static
by GitHub pull_request - opened 01 Jul '23

01 Jul '23

alsa-project/alsa-lib pull request #335 was opened from neobrain: I've ran into build issues in a project that links against both libalsa and libxkbcommon, which both export a function called `parse_string`. Changing internal function to internal linkage (`static`) prevents this kind of problem. Request URL : https://github.com/alsa-project/alsa-lib/pull/335 Patch URL : https://github.com/alsa-project/alsa-lib/pull/335.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 RESEND RESEND] ASoC: fsl MPC52xx drivers require PPC_BESTCOMM
by Randy Dunlap 01 Jul '23

01 Jul '23

Both SND_MPC52xx_SOC_PCM030 and SND_MPC52xx_SOC_EFIKA select SND_SOC_MPC5200_AC97. The latter symbol depends on PPC_BESTCOMM, so the 2 former symbols should also depend on PPC_BESTCOMM since "select" does not follow any dependency chains. This prevents a kconfig warning and build errors: WARNING: unmet direct dependencies detected for SND_SOC_MPC5200_AC97 Depends on [n]: SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_MPC52xx [=y] && PPC_BESTCOMM [=n] Selected by [m]: - SND_MPC52xx_SOC_PCM030 [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_MPC5200_SIMPLE [=y] - SND_MPC52xx_SOC_EFIKA [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_EFIKA [=y] ERROR: modpost: "mpc5200_audio_dma_destroy" [sound/soc/fsl/mpc5200_psc_ac97.ko] undefined! ERROR: modpost: "mpc5200_audio_dma_create" [sound/soc/fsl/mpc5200_psc_ac97.ko] undefined! Fixes: 40d9ec14e7e1 ("ASoC: remove BROKEN from Efika and pcm030 fabric drivers") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Shengjiu Wang <shengjiu.wang(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> --- v2: use correct email address for Mark Brown. sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff -- a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -243,7 +243,7 @@ config SND_SOC_MPC5200_AC97 config SND_MPC52xx_SOC_PCM030 tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" - depends on PPC_MPC5200_SIMPLE + depends on PPC_MPC5200_SIMPLE && PPC_BESTCOMM select SND_SOC_MPC5200_AC97 select SND_SOC_WM9712 help @@ -252,7 +252,7 @@ config SND_MPC52xx_SOC_PCM030 config SND_MPC52xx_SOC_EFIKA tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" - depends on PPC_EFIKA + depends on PPC_EFIKA && PPC_BESTCOMM select SND_SOC_MPC5200_AC97 select SND_SOC_STAC9766 help

1 0

[PATCH 0/2] ALSA: hda/realtek: Add quirk for ASUS ROG laptops
by Luke D. Jones 01 Jul '23

01 Jul '23

Adds two quirks for each GX650P and GA402X laptop with the I2C connected Cirrus amp. The added quirk chain is similar to the one added for ROG Ally but with the verbs removed. It is likely that the ROG Ally quirk will move to this set in future. Luke D. Jones (2): ALSA: hda/realtek: Add quirk for ASUS ROG GX650P ALSA: hda/realtek: Add quirk for ASUS ROG GA402X sound/pci/hda/patch_realtek.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) -- 2.41.0

3 4

[PATCH 0/7] dt-bindings: restrict node name suffixes
by Krzysztof Kozlowski 30 Jun '23

30 Jun '23

Hi, Tree-wide cleanup of DTS node name suffixes "-N", e.g. "pwm-5", so we allow only decimal numbers. In few cases narrow the pattern to also disallow multiple suffixes, e.g. "pwm-5-5". No dependencies, can be applied by individual subsystems. Cc: Tony Lindgren <tony(a)atomide.com> Cc: Oleksij Rempel <o.rempel(a)pengutronix.de> Link: https://lore.kernel.org/all/20221127182232.GA128974-robh@kernel.org/ Best regards, Krzysztof Krzysztof Kozlowski (7): dt-bindings: phy: intel,combo-phy: restrict node name suffixes dt-bindings: pwm: restrict node name suffixes dt-bindings: rtc: restrict node name suffixes dt-bindings: slimbus: restrict node name suffixes spi: dt-bindings: restrict node name suffixes dt-bindings: timestamp: restrict node name suffixes dt-bindings: watchdog: restrict node name suffixes Documentation/devicetree/bindings/phy/intel,combo-phy.yaml | 2 +- Documentation/devicetree/bindings/pwm/pwm.yaml | 2 +- Documentation/devicetree/bindings/rtc/rtc.yaml | 2 +- Documentation/devicetree/bindings/slimbus/slimbus.yaml | 2 +- Documentation/devicetree/bindings/spi/spi-controller.yaml | 2 +- .../bindings/timestamp/hardware-timestamps-common.yaml | 2 +- Documentation/devicetree/bindings/watchdog/watchdog.yaml | 4 ++-- 7 files changed, 8 insertions(+), 8 deletions(-) -- 2.34.1

10 31

[PATCH] ASoC: codecs: wcd938x: fix codec initialisation race
by Johan Hovold 30 Jun '23

30 Jun '23

Make sure to resume the codec and soundwire device before trying to read the codec variant and configure the device during component probe. This specifically avoids interpreting (a masked and shifted) -EBUSY errno as the variant: wcd938x_codec audio-codec: ASoC: error at soc_component_read_no_lock on audio-codec for register: [0x000034b0] -16 when the soundwire device happens to be suspended, which in turn prevents some headphone controls from being registered. Fixes: 8d78602aa87a ("ASoC: codecs: wcd938x: add basic driver") Cc: stable(a)vger.kernel.org # 5.14 Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reported-by: Steev Klimaszewski <steev(a)kali.org> Signed-off-by: Johan Hovold <johan+linaro(a)kernel.org> --- sound/soc/codecs/wcd938x.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index e7d6a02cdec0..e3ae4fb2c4db 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3085,6 +3085,10 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component) snd_soc_component_init_regmap(component, wcd938x->regmap); + ret = pm_runtime_resume_and_get(dev); + if (ret < 0) + return ret; + wcd938x->variant = snd_soc_component_read_field(component, WCD938X_DIGITAL_EFUSE_REG_0, WCD938X_ID_MASK); @@ -3098,6 +3102,8 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component) (WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0); } + pm_runtime_put(dev); + wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip, WCD938X_IRQ_HPHR_PDM_WD_INT); wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip, -- 2.39.3

2 1

[PATCH] ASoC: codecs: es8326: Fix power-up sequence
by Zhu Ning 30 Jun '23

30 Jun '23

Old power-up sequence causes large pop noise during start-up. Using a new sequence instead. Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 112 +++++++++++++++++++++++++------------- sound/soc/codecs/es8326.h | 14 +++-- 2 files changed, 83 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 9190a8779334..0b702bdf033a 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -159,7 +159,7 @@ static const struct snd_soc_dapm_route es8326_dapm_routes[] = { }; static const struct regmap_range es8326_volatile_ranges[] = { - regmap_reg_range(ES8326_HP_DETECT, ES8326_HP_DETECT), + regmap_reg_range(ES8326_CTIA_OMTP_STA, ES8326_CSM_MUTE_STA), }; static const struct regmap_access_table es8326_volatile_table = { @@ -415,15 +415,17 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, ret = clk_prepare_enable(es8326->mclk); if (ret) return ret; - regmap_write(es8326->regmap, ES8326_RESET, ES8326_PWRUP_SEQ_EN); - regmap_write(es8326->regmap, ES8326_INTOUT_IO, 0x45); + + regmap_write(es8326->regmap, ES8326_RESET, 0x9f); + msleep(20); + regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x00); + regmap_write(es8326->regmap, ES8326_INTOUT_IO, es8326->interrupt_clk); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); - regmap_write(es8326->regmap, ES8326_SDINOUT23_IO, ES8326_IO_INPUT); - regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE, 0x05); - regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x02); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E); regmap_write(es8326->regmap, ES8326_PGA_PDN, 0x40); - regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0xAA); + regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); + regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_ON); break; case SND_SOC_BIAS_PREPARE: @@ -432,18 +434,15 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, break; case SND_SOC_BIAS_OFF: clk_disable_unprepare(es8326->mclk); - regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x11); regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_OFF); - regmap_write(es8326->regmap, ES8326_PGA_PDN, 0xF8); + regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x00); - regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x08); + regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); - regmap_write(es8326->regmap, ES8326_SDINOUT23_IO, ES8326_IO_INPUT); regmap_write(es8326->regmap, ES8326_RESET, ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); break; } - return 0; } @@ -519,7 +518,7 @@ static void es8326_jack_button_handler(struct work_struct *work) return; mutex_lock(&es8326->lock); - iface = snd_soc_component_read(comp, ES8326_HP_DETECT); + iface = snd_soc_component_read(comp, ES8326_HPDET_STA); switch (iface) { case 0x93: /* pause button detected */ @@ -578,7 +577,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) unsigned int iface; mutex_lock(&es8326->lock); - iface = snd_soc_component_read(comp, ES8326_HP_DETECT); + iface = snd_soc_component_read(comp, ES8326_HPDET_STA); dev_dbg(comp->dev, "gpio flag %#04x", iface); if ((iface & ES8326_HPINSERT_FLAG) == 0) { /* Jack unplugged or spurious IRQ */ @@ -638,36 +637,73 @@ static int es8326_resume(struct snd_soc_component *component) regcache_cache_only(es8326->regmap, false); regcache_sync(es8326->regmap); + /* reset register value to default */ + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); + usleep_range(1000, 3000); + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); + /* reset internal state */ + regmap_write(es8326->regmap, ES8326_RESET, 0x1f); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0x88); + /* set headphone default type and detect pin */ + regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x81 | + ES8326_HP_DET_SRC_PIN9); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE, 0x05); + + /* set internal oscillator as clock source of headpone cp */ + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC, 0x84); regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_ON); - /* Two channel ADC */ + /* clock manager reset release */ + regmap_write(es8326->regmap, ES8326_RESET, 0x17); + /* set headphone detection as half scan mode */ + regmap_write(es8326->regmap, ES8326_HP_MISC, 0x08); regmap_write(es8326->regmap, ES8326_PULLUP_CTL, 0x02); + + /* enable headphone driver */ + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xa7); + regmap_write(es8326->regmap, ES8326_HP_DRIVER_REF, 0xa3); + regmap_write(es8326->regmap, ES8326_HP_DRIVER_REF, 0xb3); + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xa1); + regmap_write(es8326->regmap, ES8326_CLK_INV, 0x00); - regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC, 0x1F); - regmap_write(es8326->regmap, ES8326_CLK_VMIDS1, 0xC8); - regmap_write(es8326->regmap, ES8326_CLK_VMIDS2, 0x88); - regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME, 0x20); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1, 0xc4); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2, 0x81); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME, 0x00); + + /* turn off headphone out */ + regmap_write(es8326->regmap, ES8326_HP_CAL, 0x00); + /* set ADC and DAC in low power mode */ + regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0); + + /* force micbias on */ + regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0x4f); regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x08); - regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x22); - regmap_write(es8326->regmap, ES8326_ADC1_SRC, es8326->mic1_src); - regmap_write(es8326->regmap, ES8326_ADC2_SRC, es8326->mic2_src); - regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0x88); - regmap_write(es8326->regmap, ES8326_HP_DET, - ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); - regmap_write(es8326->regmap, ES8326_INT_SOURCE, es8326->interrupt_src); - regmap_write(es8326->regmap, ES8326_INTOUT_IO, es8326->interrupt_clk); + regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7F); + /* select vdda as micbias source */ + regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x23); + /* set dac dsmclip = 1 */ + regmap_write(es8326->regmap, ES8326_DAC_DSM, 0x08); + regmap_write(es8326->regmap, ES8326_DAC_VPPSCALE, 0x15); + + regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x0c); + regmap_write(es8326->regmap, ES8326_INTOUT_IO, + es8326->interrupt_clk); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO, ES8326_IO_INPUT); + + regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_ON); - snd_soc_component_update_bits(component, ES8326_PGAGAIN, - ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + regmap_update_bits(es8326->regmap, ES8326_PGAGAIN, ES8326_MIC_SEL_MASK, + ES8326_MIC1_SEL); - regmap_read(es8326->regmap, ES8326_CHIP_VERSION, &reg); - if ((reg & ES8326_VERSION_B) == 1) { - regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xDD); - regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7F); - regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x0F); - /* enable button detect */ - regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xA0); - } + regmap_update_bits(es8326->regmap, ES8326_DAC_MUTE, ES8326_MUTE_MASK, + ES8326_MUTE); + regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x80 | + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_read(es8326->regmap, ES8326_CHIP_VERSION, &reg); + es8326->version = reg; es8326_irq(es8326->irq, es8326); return 0; } @@ -743,7 +779,7 @@ static void es8326_enable_jack_detect(struct snd_soc_component *component, mutex_lock(&es8326->lock); if (es8326->jd_inverted) - snd_soc_component_update_bits(component, ES8326_HP_DET, + snd_soc_component_update_bits(component, ES8326_HPDET_TYPE, ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); es8326->jack = jack; diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h index 1eb9d0095378..9093222ae04d 100644 --- a/sound/soc/codecs/es8326.h +++ b/sound/soc/codecs/es8326.h @@ -9,8 +9,6 @@ #ifndef _ES8326_H #define _ES8326_H -#define CONFIG_HHTECH_MINIPMP 1 - /* ES8326 register space */ #define ES8326_RESET 0x00 #define ES8326_CLK_CTL 0x01 @@ -75,15 +73,19 @@ #define ES8326_DRC_RECOVERY 0x53 #define ES8326_DRC_WINSIZE 0x54 #define ES8326_HPJACK_TIMER 0x56 -#define ES8326_HP_DET 0x57 +#define ES8326_HPDET_TYPE 0x57 #define ES8326_INT_SOURCE 0x58 #define ES8326_INTOUT_IO 0x59 #define ES8326_SDINOUT1_IO 0x5A #define ES8326_SDINOUT23_IO 0x5B #define ES8326_JACK_PULSE 0x5C +#define ES8326_HP_MISC 0xF7 +#define ES8326_CTIA_OMTP_STA 0xF8 #define ES8326_PULLUP_CTL 0xF9 -#define ES8326_HP_DETECT 0xFB +#define ES8326_CSM_I2C_STA 0xFA +#define ES8326_HPDET_STA 0xFB +#define ES8326_CSM_MUTE_STA 0xFC #define ES8326_CHIP_ID1 0xFD #define ES8326_CHIP_ID2 0xFE #define ES8326_CHIP_VERSION 0xFF @@ -179,6 +181,8 @@ #define ES8326_HPBUTTON_FLAG (1 << 0) /* ES8326_CHIP_VERSION 0xFF */ -#define ES8326_VERSION_B (1 << 0) +#define ES8326_VERSION (1 << 0) +#define ES8326_VERSION_B (3 << 0) #endif + -- 2.34.1

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[tiwai-sound:topic/midi20-gadget 58/62] f_midi2.c:undefined reference to `snd_ump_switch_protocol'
by kernel test robot 30 Jun '23

30 Jun '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20-gadget head: 9d606002a91a37796ae62afc5e6667126314e21b commit: 0417de91e87de93793cd0d2cc4aef9062dc62f05 [58/62] usb: gadget: Add support for USB MIDI 2.0 function driver config: x86_64-buildonly-randconfig-r001-20230629 (https://download.01.org/0day-ci/archive/20230630/202306301830.QG614gSO-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce: (https://download.01.org/0day-ci/archive/20230630/202306301830.QG614gSO-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202306301830.QG614gSO-lkp@intel.com/ All errors (new ones prefixed by >>): ld: drivers/usb/gadget/function/f_midi2.o: in function `f_midi2_create_card.constprop.0': f_midi2.c:(.text+0x35b6): undefined reference to `snd_ump_endpoint_new' ld: f_midi2.c:(.text+0x3bba): undefined reference to `snd_ump_block_new' ld: drivers/usb/gadget/function/f_midi2.o: in function `process_ump_stream_msg': >> f_midi2.c:(.text+0x4b0f): undefined reference to `snd_ump_switch_protocol' ld: drivers/usb/gadget/function/f_midi2.o: in function `f_midi2_ep_out_complete': f_midi2.c:(.text+0x598e): undefined reference to `snd_ump_receive_ump_val' ld: f_midi2.c:(.text+0x5b36): undefined reference to `snd_ump_receive' ld: drivers/usb/gadget/function/f_midi2.o: in function `process_ump_transmit': f_midi2.c:(.text+0x5de1): undefined reference to `snd_ump_transmit' -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[tiwai-sound:topic/midi20-gadget 58/62] ERROR: modpost: "snd_ump_switch_protocol" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined!
by kernel test robot 30 Jun '23

30 Jun '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20-gadget head: 9d606002a91a37796ae62afc5e6667126314e21b commit: 0417de91e87de93793cd0d2cc4aef9062dc62f05 [58/62] usb: gadget: Add support for USB MIDI 2.0 function driver config: openrisc-randconfig-r001-20230630 (https://download.01.org/0day-ci/archive/20230630/202306301823.km1gb5Cd-lkp@…) compiler: or1k-linux-gcc (GCC) 12.3.0 reproduce: (https://download.01.org/0day-ci/archive/20230630/202306301823.km1gb5Cd-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202306301823.km1gb5Cd-lkp@intel.com/ All errors (new ones prefixed by >>, old ones prefixed by <<): WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_ext_maps (section: .data) -> qed_mfw_ext_1g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_ext_maps (section: .data) -> qed_mfw_ext_10g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_ext_maps (section: .data) -> qed_mfw_ext_25g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_ext_maps (section: .data) -> qed_mfw_ext_40g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_ext_maps (section: .data) -> qed_mfw_ext_50g_base_r (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_ext_maps (section: .data) -> qed_mfw_ext_50g_base_r2 (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_ext_maps (section: .data) -> qed_mfw_ext_100g_base_r2 (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_ext_maps (section: .data) -> qed_mfw_ext_100g_base_r4 (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_legacy_maps (section: .data) -> qed_mfw_legacy_1g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_legacy_maps (section: .data) -> qed_mfw_legacy_10g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_legacy_maps (section: .data) -> qed_mfw_legacy_20g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_legacy_maps (section: .data) -> qed_mfw_legacy_25g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_legacy_maps (section: .data) -> qed_mfw_legacy_40g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_legacy_maps (section: .data) -> qed_mfw_legacy_50g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qed/qed.o: section mismatch in reference: qed_mfw_legacy_maps (section: .data) -> qed_mfw_legacy_bb_100g (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qede/qede.o: section mismatch in reference: qede_forced_speed_maps (section: .data) -> qede_forced_speed_1000 (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qede/qede.o: section mismatch in reference: qede_forced_speed_maps (section: .data) -> qede_forced_speed_10000 (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qede/qede.o: section mismatch in reference: qede_forced_speed_maps (section: .data) -> qede_forced_speed_20000 (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qede/qede.o: section mismatch in reference: qede_forced_speed_maps (section: .data) -> qede_forced_speed_25000 (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qede/qede.o: section mismatch in reference: qede_forced_speed_maps (section: .data) -> qede_forced_speed_40000 (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qede/qede.o: section mismatch in reference: qede_forced_speed_maps (section: .data) -> qede_forced_speed_50000 (section: .init.rodata) WARNING: modpost: drivers/net/ethernet/qlogic/qede/qede.o: section mismatch in reference: qede_forced_speed_maps (section: .data) -> qede_forced_speed_100000 (section: .init.rodata) ERROR: modpost: "snd_ump_block_new" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! >> ERROR: modpost: "snd_ump_switch_protocol" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! ERROR: modpost: "snd_ump_receive_ump_val" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! ERROR: modpost: "snd_ump_transmit" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! ERROR: modpost: "snd_ump_receive" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! ERROR: modpost: "snd_ump_endpoint_new" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[PATCH 0/3] ASoC: SOF: pcm/Intel: Handle IPC dependent sequencing correctly
by Peter Ujfalusi 30 Jun '23

30 Jun '23

Hi, IPC3 and IPC4 firmwares handle and execute tasks at different stages, like managing DMAs. In most cases these are aligned, but we have few exceptions that needs to be handled differently. This series introduces flags to handle the differing cases to make sure that the correct sequencing is used regerless of the IPC version. Regards, Peter --- Ranjani Sridharan (3): ASoC: SOF: Intel: hda-dai: Do not perform DMA cleanup during stop ASoC: SOF: pcm: Make hw_params reset conditional for IPC3 ASoC: SOF: pcm: Improve the pcm trigger sequence sound/soc/sof/intel/hda-dai.c | 1 - sound/soc/sof/ipc3-pcm.c | 1 + sound/soc/sof/ipc4-pcm.c | 3 ++- sound/soc/sof/pcm.c | 29 ++++++++++++++++++++++------- sound/soc/sof/sof-audio.h | 6 ++++++ 5 files changed, 31 insertions(+), 9 deletions(-) -- 2.40.0

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[PATCH v2 0/2] ASoC: fsl_sai: Fill Tx FIFO to avoid initial underruns
by Sascha Hauer 30 Jun '23

30 Jun '23

This series fixes initial underruns that can occur in the TX queue of the fsl_sai interface when starting playback. These patches are around here for quite some time and have proven useful. Time to upstream them. Sascha Changes since v1: - Add missing Signed-off-by: Ahmad Fatoum (2): ASoC: fsl_sai: refactor TDM slots calculation into helper function ASoC: fsl_sai: Fill Tx FIFO to avoid initial underruns sound/soc/fsl/fsl_sai.c | 38 +++++++++++++++++++++++++++++++++----- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 34 insertions(+), 5 deletions(-) -- 2.39.2

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[PATCH] ASoC: core: Always store of_node when getting DAI link component
by Mark Brown 30 Jun '23

30 Jun '23

The generic snd_soc_dai_get_dlc() contains a default translation function for DAI names which has factored out common code in a number of card drivers, resolving the dai_name and of_node either using a driver provided translation function or with a generic implementation. Unfortunately the of_node can't be set by the translation function since it currently doesn't have an interface to do that but snd_soc_dai_get_dlc() only initialises the of_node in the case where there is no translation function. This breaks the Meson support after conversion to use the generic helpers since the DPCM cards for it check which component of the SoC is connected to each link by checking the compatible for the component and the Meson components provide a custom operation so don't use the generic code. Fix this and potentially other cards by unconditionally storing the node in the dai_link_component, there shouldn't be a binding specific of_node selected since that's how we determine the translation function. Fixes: 2e1dbea1f8a3 ("ASoC: meson: use snd_soc_{of_}get_dlc()") Fixes: 3c8b5861850c ("ASoC: soc-core.c: add index on snd_soc_of_get_dai_name()") Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f06a20773a34..11bc5250ffd0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3262,6 +3262,8 @@ int snd_soc_get_dlc(const struct of_phandle_args *args, struct snd_soc_dai_link_ struct snd_soc_component *pos; int ret = -EPROBE_DEFER; + dlc->of_node = args->np; + mutex_lock(&client_mutex); for_each_component(pos) { struct device_node *component_of_node = soc_component_to_node(pos); @@ -3300,7 +3302,6 @@ int snd_soc_get_dlc(const struct of_phandle_args *args, struct snd_soc_dai_link_ id--; } - dlc->of_node = args->np; dlc->dai_name = dai->driver->name; if (!dlc->dai_name) dlc->dai_name = pos->name; --- base-commit: 154756319cc6f8b8b86241da02da6a8fcc6abd1f change-id: 20230623-asoc-fix-meson-probe-94ecd113bebb Best regards, -- Mark Brown,,, <broonie(a)kernel.org>

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[tiwai-sound:topic/midi20-gadget 56/56] f_midi2.c:undefined reference to `snd_ump_endpoint_new'
by kernel test robot 30 Jun '23

30 Jun '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20-gadget head: ed28b6939ebadd073e731ccf0f2c6ce4ab102ffd commit: ed28b6939ebadd073e731ccf0f2c6ce4ab102ffd [56/56] usb: gadget: add support for USB MIDI 2.0 function driver (WIP) config: x86_64-buildonly-randconfig-r001-20230629 (https://download.01.org/0day-ci/archive/20230630/202306300642.o48bLzoB-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce: (https://download.01.org/0day-ci/archive/20230630/202306300642.o48bLzoB-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202306300642.o48bLzoB-lkp@intel.com/ All errors (new ones prefixed by >>): ld: drivers/usb/gadget/function/f_midi2.o: in function `f_midi2_create.constprop.0': >> f_midi2.c:(.text+0x27f2): undefined reference to `snd_ump_endpoint_new' >> ld: f_midi2.c:(.text+0x29a5): undefined reference to `snd_ump_block_new' ld: drivers/usb/gadget/function/f_midi2.o: in function `process_ump_transmit': >> f_midi2.c:(.text+0x4a69): undefined reference to `snd_ump_transmit' ld: drivers/usb/gadget/function/f_midi2.o: in function `f_midi2_ep_out_complete': >> f_midi2.c:(.text+0x5af1): undefined reference to `snd_ump_receive_ump_val' >> ld: f_midi2.c:(.text+0x5bef): undefined reference to `snd_ump_receive' -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[tiwai-sound:topic/midi20-gadget 56/56] ld.lld: error: undefined symbol: snd_ump_transmit
by kernel test robot 30 Jun '23

30 Jun '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20-gadget head: ed28b6939ebadd073e731ccf0f2c6ce4ab102ffd commit: ed28b6939ebadd073e731ccf0f2c6ce4ab102ffd [56/56] usb: gadget: add support for USB MIDI 2.0 function driver (WIP) config: i386-randconfig-r023-20230629 (https://download.01.org/0day-ci/archive/20230630/202306300550.yyh6cWzj-lkp@…) compiler: clang version 15.0.7 (https://github.com/llvm/llvm-project.git 8dfdcc7b7bf66834a761bd8de445840ef68e4d1a) reproduce: (https://download.01.org/0day-ci/archive/20230630/202306300550.yyh6cWzj-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202306300550.yyh6cWzj-lkp@intel.com/ All errors (new ones prefixed by >>): >> ld.lld: error: undefined symbol: snd_ump_transmit >>> referenced by f_midi2.c:644 (drivers/usb/gadget/function/f_midi2.c:644) >>> drivers/usb/gadget/function/f_midi2.o:(process_ump_transmit) in archive vmlinux.a -- >> ld.lld: error: undefined symbol: snd_ump_endpoint_new >>> referenced by f_midi2.c:946 (drivers/usb/gadget/function/f_midi2.c:946) >>> drivers/usb/gadget/function/f_midi2.o:(f_midi2_bind) in archive vmlinux.a -- >> ld.lld: error: undefined symbol: snd_ump_block_new >>> referenced by f_midi2.c:957 (drivers/usb/gadget/function/f_midi2.c:957) >>> drivers/usb/gadget/function/f_midi2.o:(f_midi2_bind) in archive vmlinux.a -- >> ld.lld: error: undefined symbol: snd_ump_receive >>> referenced by f_midi2.c:623 (drivers/usb/gadget/function/f_midi2.c:623) >>> drivers/usb/gadget/function/f_midi2.o:(f_midi2_ep_out_complete) in archive vmlinux.a -- >> ld.lld: error: undefined symbol: snd_ump_receive_ump_val >>> referenced by f_midi2.c:599 (drivers/usb/gadget/function/f_midi2.c:599) >>> drivers/usb/gadget/function/f_midi2.o:(f_midi2_ep_out_complete) in archive vmlinux.a -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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configure: Add large file support via CFLAGS
by GitHub pull_request - opened 30 Jun '23

30 Jun '23

alsa-project/alsa-utils pull request #223 was opened from kendase3: Currently scanelf (pax-utils) detects missing LFS support in the following files/calls: __open_2,fstat,mmap,open,scandir,stat,alphasort,lstat /usr/sbin/alsactl fopen /usr/lib/alsa-topology/libalsatplg_module_nhlt.so fopen /usr/bin/alsaucm fopen /usr/bin/aplaymidi __open_2,open /usr/bin/alsatplg fstat,open,__open_2 /usr/bin/axfer fopen /usr/bin/arecordmidi fopen,__open_2 /usr/bin/speaker-test creat,__open_2,lseek /usr/bin/amidi Note that this may cause problems on systems with a 32-bit kernel, but I've tested playing audio on a more recent 32-bit system with a 64-bit kernel. Let me know if you'd prefer a different approach! Notably the existing AC_SYS_LARGEFILE in configure.ac was not having any effect in my environment. I'm building on a 64-bit host for a 32-bit target. Request URL : https://github.com/alsa-project/alsa-utils/pull/223 Patch URL : https://github.com/alsa-project/alsa-utils/pull/223.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[tiwai-sound:topic/midi20-gadget 56/56] ERROR: modpost: "snd_ump_block_new" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined!
by kernel test robot 29 Jun '23

29 Jun '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20-gadget head: ed28b6939ebadd073e731ccf0f2c6ce4ab102ffd commit: ed28b6939ebadd073e731ccf0f2c6ce4ab102ffd [56/56] usb: gadget: add support for USB MIDI 2.0 function driver (WIP) config: arm-randconfig-r046-20230629 (https://download.01.org/0day-ci/archive/20230630/202306300438.HICMJktb-lkp@…) compiler: arm-linux-gnueabi-gcc (GCC) 12.3.0 reproduce: (https://download.01.org/0day-ci/archive/20230630/202306300438.HICMJktb-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202306300438.HICMJktb-lkp@intel.com/ All error/warnings (new ones prefixed by >>, old ones prefixed by <<): >> drivers/usb/gadget/function/f_midi2.c:483:6: warning: no previous prototype for 'reply_ump_stream_fb_info' [-Wmissing-prototypes] 483 | void reply_ump_stream_fb_info(struct f_midi2 *midi2, int blk) | ^~~~~~~~~~~~~~~~~~~~~~~~ -- >> ERROR: modpost: "snd_ump_block_new" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! >> ERROR: modpost: "snd_ump_receive_ump_val" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! >> ERROR: modpost: "snd_ump_transmit" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! >> ERROR: modpost: "snd_ump_receive" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! >> ERROR: modpost: "snd_ump_endpoint_new" [drivers/usb/gadget/function/usb_f_midi2.ko] undefined! -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[tiwai-sound:topic/midi20-gadget 56/56] drivers/usb/gadget/function/f_midi2.c:483:6: warning: no previous prototype for function 'reply_ump_stream_fb_info'
by kernel test robot 29 Jun '23

29 Jun '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20-gadget head: ed28b6939ebadd073e731ccf0f2c6ce4ab102ffd commit: ed28b6939ebadd073e731ccf0f2c6ce4ab102ffd [56/56] usb: gadget: add support for USB MIDI 2.0 function driver (WIP) config: i386-randconfig-r023-20230629 (https://download.01.org/0day-ci/archive/20230630/202306300419.dOOgnkIH-lkp@…) compiler: clang version 15.0.7 (https://github.com/llvm/llvm-project.git 8dfdcc7b7bf66834a761bd8de445840ef68e4d1a) reproduce: (https://download.01.org/0day-ci/archive/20230630/202306300419.dOOgnkIH-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202306300419.dOOgnkIH-lkp@intel.com/ All warnings (new ones prefixed by >>): >> drivers/usb/gadget/function/f_midi2.c:483:6: warning: no previous prototype for function 'reply_ump_stream_fb_info' [-Wmissing-prototypes] void reply_ump_stream_fb_info(struct f_midi2 *midi2, int blk) ^ drivers/usb/gadget/function/f_midi2.c:483:1: note: declare 'static' if the function is not intended to be used outside of this translation unit void reply_ump_stream_fb_info(struct f_midi2 *midi2, int blk) ^ static 1 warning generated. vim +/reply_ump_stream_fb_info +483 drivers/usb/gadget/function/f_midi2.c 481 482 /* exported for shell */ > 483 void reply_ump_stream_fb_info(struct f_midi2 *midi2, int blk) 484 { 485 u32 buf[4]; 486 487 memset(buf, 0, sizeof(buf)); 488 buf[0] = ump_stream_compose(UMP_STREAM_MSG_STATUS_FB_INFO, 0); 489 buf[0] |= (midi2->blks[blk].active << 15) | (blk << 8); 490 buf[0] |= midi2->blks[blk].dir << 4; // UI-hint 491 buf[1] |= 0 << 2; // MIDI 1.0 port? No 492 buf[0] |= midi2->blks[blk].dir; // direction 493 buf[1] |= (midi2->blks[blk].start_group << 24) | // start group 494 (midi2->blks[blk].num_groups << 16); // # groups spanned 495 buf[1] |= 0x01 << 8; // MIDI-CI support version 496 buf[1] |= 0; // Sysex8 streams; not supported 497 498 reply_ep_in(midi2, buf, sizeof(buf)); 499 } 500 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[tiwai-sound:topic/midi20-gadget 56/56] drivers/usb/gadget/function/f_midi2.c:483:6: warning: no previous prototype for 'reply_ump_stream_fb_info'
by kernel test robot 29 Jun '23

29 Jun '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20-gadget head: ed28b6939ebadd073e731ccf0f2c6ce4ab102ffd commit: ed28b6939ebadd073e731ccf0f2c6ce4ab102ffd [56/56] usb: gadget: add support for USB MIDI 2.0 function driver (WIP) config: arc-randconfig-r005-20230629 (https://download.01.org/0day-ci/archive/20230630/202306300411.zojb9GzL-lkp@…) compiler: arceb-elf-gcc (GCC) 12.3.0 reproduce: (https://download.01.org/0day-ci/archive/20230630/202306300411.zojb9GzL-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202306300411.zojb9GzL-lkp@intel.com/ All warnings (new ones prefixed by >>): >> drivers/usb/gadget/function/f_midi2.c:483:6: warning: no previous prototype for 'reply_ump_stream_fb_info' [-Wmissing-prototypes] 483 | void reply_ump_stream_fb_info(struct f_midi2 *midi2, int blk) | ^~~~~~~~~~~~~~~~~~~~~~~~ vim +/reply_ump_stream_fb_info +483 drivers/usb/gadget/function/f_midi2.c 481 482 /* exported for shell */ > 483 void reply_ump_stream_fb_info(struct f_midi2 *midi2, int blk) 484 { 485 u32 buf[4]; 486 487 memset(buf, 0, sizeof(buf)); 488 buf[0] = ump_stream_compose(UMP_STREAM_MSG_STATUS_FB_INFO, 0); 489 buf[0] |= (midi2->blks[blk].active << 15) | (blk << 8); 490 buf[0] |= midi2->blks[blk].dir << 4; // UI-hint 491 buf[1] |= 0 << 2; // MIDI 1.0 port? No 492 buf[0] |= midi2->blks[blk].dir; // direction 493 buf[1] |= (midi2->blks[blk].start_group << 24) | // start group 494 (midi2->blks[blk].num_groups << 16); // # groups spanned 495 buf[1] |= 0x01 << 8; // MIDI-CI support version 496 buf[1] |= 0; // Sysex8 streams; not supported 497 498 reply_ep_in(midi2, buf, sizeof(buf)); 499 } 500 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[PATCH AUTOSEL 4.14 1/4] ASoC: ops: Check bounds for second channel in snd_soc_put_volsw_sx()
by Sasha Levin 29 Jun '23

29 Jun '23

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 97eea946b93961fffd29448dcda7398d0d51c4b2 ] The bounds checks in snd_soc_put_volsw_sx() are only being applied to the first channel, meaning it is possible to write out of bounds values to the second channel in stereo controls. Add appropriate checks. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220511134137.169575-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 81c9ecfa7c7f9..63c0e61b17541 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -465,6 +465,12 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, if (snd_soc_volsw_is_stereo(mc)) { val_mask = mask << rshift; val2 = (ucontrol->value.integer.value[1] + min) & mask; + + if (mc->platform_max && val2 > mc->platform_max) + return -EINVAL; + if (val2 > max) + return -EINVAL; + val2 = val2 << rshift; err = snd_soc_component_update_bits(component, reg2, val_mask, -- 2.39.2

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[PATCH AUTOSEL 4.19 1/5] ASoC: ops: Check bounds for second channel in snd_soc_put_volsw_sx()
by Sasha Levin 29 Jun '23

29 Jun '23

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 97eea946b93961fffd29448dcda7398d0d51c4b2 ] The bounds checks in snd_soc_put_volsw_sx() are only being applied to the first channel, meaning it is possible to write out of bounds values to the second channel in stereo controls. Add appropriate checks. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220511134137.169575-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 453b61b42dd9e..00e6a6e46fe52 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -460,6 +460,12 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, if (snd_soc_volsw_is_stereo(mc)) { val_mask = mask << rshift; val2 = (ucontrol->value.integer.value[1] + min) & mask; + + if (mc->platform_max && val2 > mc->platform_max) + return -EINVAL; + if (val2 > max) + return -EINVAL; + val2 = val2 << rshift; err = snd_soc_component_update_bits(component, reg2, val_mask, -- 2.39.2

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[PATCH AUTOSEL 5.4 1/5] ASoC: ops: Check bounds for second channel in snd_soc_put_volsw_sx()
by Sasha Levin 29 Jun '23

29 Jun '23

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 97eea946b93961fffd29448dcda7398d0d51c4b2 ] The bounds checks in snd_soc_put_volsw_sx() are only being applied to the first channel, meaning it is possible to write out of bounds values to the second channel in stereo controls. Add appropriate checks. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220511134137.169575-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 453b61b42dd9e..00e6a6e46fe52 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -460,6 +460,12 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, if (snd_soc_volsw_is_stereo(mc)) { val_mask = mask << rshift; val2 = (ucontrol->value.integer.value[1] + min) & mask; + + if (mc->platform_max && val2 > mc->platform_max) + return -EINVAL; + if (val2 > max) + return -EINVAL; + val2 = val2 << rshift; err = snd_soc_component_update_bits(component, reg2, val_mask, -- 2.39.2

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[PATCH AUTOSEL 6.1 11/12] ALSA: hda/realtek: Add quirk for ASUS ROG GV601V
by Sasha Levin 29 Jun '23

29 Jun '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 82edd1bd7f98567928871e2a2a317724d35f0085 ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG GV601V series. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230621085715.5382-1-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 3889701bdb1c9..0547f60a400e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9527,6 +9527,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), -- 2.39.2

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[PATCH AUTOSEL 6.1 07/12] ALSA: hda/realtek: Add quirk for ASUS ROG G634Z
by Sasha Levin 29 Jun '23

29 Jun '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 555434fd5c6b3589d9511ab6e88faf50346e19da ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG G634Z series. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230619060320.1336455-1-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 920e44ba998a5..3889701bdb1c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9552,6 +9552,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), + SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 16/17] ALSA: hda/realtek: Add quirk for ASUS ROG GV601V
by Sasha Levin 29 Jun '23

29 Jun '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 82edd1bd7f98567928871e2a2a317724d35f0085 ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG GV601V series. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230621085715.5382-1-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 3889701bdb1c9..0547f60a400e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9527,6 +9527,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), -- 2.39.2

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[PATCH AUTOSEL 6.3 10/17] ALSA: hda/realtek: Add quirk for ASUS ROG G634Z
by Sasha Levin 29 Jun '23

29 Jun '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 555434fd5c6b3589d9511ab6e88faf50346e19da ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG G634Z series. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230619060320.1336455-1-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 920e44ba998a5..3889701bdb1c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9552,6 +9552,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), + SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), -- 2.39.2

1 0

[PATCH v1 0/3] ASoC: rt5677: Refactor to use device_get_match_data()
by Andy Shevchenko 29 Jun '23

29 Jun '23

The code can be simplified with device_get_match_data(). Besides that couple of additional changes, one for maintenance (patch 3) and one for making IRQ domain agnostic (not pinned to OF). Andy Shevchenko (3): ASoC: rt5677: Use agnostic irq_domain_create_linear() ASoC: rt5677: Use device_get_match_data() ASoC: rt5677: Sort headers alphabetically sound/soc/codecs/rt5677.c | 41 +++++++++++++-------------------------- sound/soc/codecs/rt5677.h | 4 ++-- 2 files changed, 16 insertions(+), 29 deletions(-) -- 2.40.0.1.gaa8946217a0b

3 7

[RFC PATCH 0/8] PCI: Define Intel PCI IDs and use them in drivers
by Amadeusz Sławiński 29 Jun '23

29 Jun '23

PCI IDs for Intel HDA are duplicated across quite a few drivers, due to various configurations and historical reasons. Currently almost all uses of HDA PCI IDs have corresponding comment telling which platform it is. Additionally there are some inconsistencies between drivers about which ID corresponds to which device. Simplify things, by adding PCI IDs to global header and make use of them in drivers. This allows for removal of comments by having IDs themselves being self explanatory. Additionally it allows for removal of existing inconsistencies by having one source of truth. Amadeusz Sławiński (8): PCI: Add Intel Audio DSP devices to pci_ids.h ALSA: intel-dsp-config: Update PCI ID list ALSA: hda: Update PCI ID list ALSA: hda/i915: Update PCI IDs ASoC: Intel: avs: Update PCI ID list ASoC: Intel: avs: Update PCI ID list ASoC: Intel: Skylake: Update PCI ID list ASoC: SOF: Intel: Update PCI ID list include/linux/pci_ids.h | 44 ++++ sound/hda/hdac_i915.c | 8 +- sound/hda/intel-dsp-config.c | 119 +++++----- sound/pci/hda/hda_intel.c | 292 ++++++++++++------------- sound/soc/intel/avs/board_selection.c | 10 +- sound/soc/intel/avs/core.c | 16 +- sound/soc/intel/skylake/skl-messages.c | 16 +- sound/soc/intel/skylake/skl.c | 36 +-- sound/soc/sof/intel/pci-apl.c | 9 +- sound/soc/sof/intel/pci-cnl.c | 15 +- sound/soc/sof/intel/pci-icl.c | 12 +- sound/soc/sof/intel/pci-mtl.c | 3 +- sound/soc/sof/intel/pci-skl.c | 6 +- sound/soc/sof/intel/pci-tgl.c | 45 ++-- sound/soc/sof/intel/pci-tng.c | 3 +- 15 files changed, 311 insertions(+), 323 deletions(-) -- 2.34.1

5 25

[PATCH 0/2] ASoC: fsl_sai: Fill Tx FIFO to avoid initial underruns
by Sascha Hauer 29 Jun '23

29 Jun '23

This series fixes initial underruns that can occur in the TX queue of the fsl_sai interface when starting playback. These patches are around here for quite some time and have proven useful. Time to upstream them. Sascha Ahmad Fatoum (2): ASoC: fsl_sai: refactor TDM slots calculation into helper function ASoC: fsl_sai: Fill Tx FIFO to avoid initial underruns sound/soc/fsl/fsl_sai.c | 38 +++++++++++++++++++++++++++++++++----- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 34 insertions(+), 5 deletions(-) -- 2.39.2

2 3

[PATCHv2] ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook G10
by Huang Rui 29 Jun '23

29 Jun '23

In the commit 3e10f6ca76c4 ("ALSA: hda/realtek: Add quirk for HP EliteBook G10 laptops"), speakers are fixed but mute/micmute LEDs does not lit. Need the ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED to get all mute/micmute LEDs working. Signed-off-by: Huang Rui <vowstar(a)gmail.com> Fixes: 3e10f6ca76c4 ("ALSA: hda/realtek: Add quirk for HP EliteBook G10 laptops") --- V1 -> V2: Add fixes to link a previously committed issue --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dabfdecece26..425c87733e79 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9491,7 +9491,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b65, "HP ProBook 455 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b66, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b70, "HP EliteBook 835 G10", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x103c, 0x8b72, "HP EliteBook 845 G10", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8b72, "HP EliteBook 845 G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b74, "HP EliteBook 845W G10", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8b77, "HP ElieBook 865 G10", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), -- 2.41.0

2 1

Arturia Minifuse 2 (and probably any other using SplitPCM macro) - lost sample rate range on each in/out (and forced to 48000)
by GitHub issues - edited 29 Jun '23

29 Jun '23

alsa-project/alsa-ucm-conf issue #333 was edited from kmarty: Hi, I have Arturia MiniFuse 2 card allowing playback/capture sample rate in range 44100 - 192000. But using channel name (e.g. `Mic/Line/Inst 1 (L)`, `Main Output L/R`) created by UCM, it offers 48000 only (and in case of playback, it offers S15_LE only). Looks like something is lost when `SplitPCM` macro is used or so (I have absolutely no idea what's going on here). Is there any possibility to make it work as it should? ================================= `/proc/asound/card4/stream0` output: ``` ARTURIA MiniFuse 2 at usb-0000:16:00.0-6.2, high speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S32_LE Channels: 4 Endpoint: 0x01 (1 OUT) (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE Sync Endpoint: 0x81 (1 IN) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: No Interface 1 Altset 2 Format: S16_LE Channels: 4 Endpoint: 0x01 (1 OUT) (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 16 Channel map: FL FR FC LFE Sync Endpoint: 0x81 (1 IN) Sync EP Interface: 1 Sync EP Altset: 2 Implicit Feedback Mode: No Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 4 Endpoint: 0x82 (2 IN) (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE ``` `alsaucm -c hw:4 dump text` output: ``` Verb.HiFi { Comment "Default Alsa Profile" Device.Line1 { Comment "Main Output L/R" Values { CaptureCTL "_ucm0001.hw:M2" PlaybackCTL "_ucm0001.hw:M2" PlaybackChannels 2 PlaybackPCM "_ucm0001.minifuse12_stereo_out:M2,0,0,1" PlaybackPriority 300 } } Device.Line2 { Comment "Loopback L/R" Values { CaptureCTL "_ucm0001.hw:M2" CaptureChannels 2 CapturePCM "_ucm0001.minifuse12_stereo_in:M2,0,2,3" CapturePriority 200 PlaybackCTL "_ucm0001.hw:M2" PlaybackChannels 2 PlaybackPCM "_ucm0001.minifuse12_stereo_out:M2,0,2,3" PlaybackPriority 200 } } Device.Line3 { Comment "Stereo Input 1+2 L/R" ConflictingDevices [ Mic2, Mic1 ] Values { CaptureCTL "_ucm0001.hw:M2" CaptureChannels 2 CapturePCM "_ucm0001.minifuse12_stereo_in:M2,0,0,1" CapturePriority 100 PlaybackCTL "_ucm0001.hw:M2" } } Device.Mic1 { Comment "Mic/Line/Inst 1 (L)" ConflictingDevices [ Line3 ] Values { CaptureCTL "_ucm0001.hw:M2" CaptureChannels 1 CapturePCM "_ucm0001.minifuse12_mono_in:M2,0,0" CapturePriority 400 PlaybackCTL "_ucm0001.hw:M2" } } Device.Mic2 { Comment "Mic/Line/Inst 2 (R)" ConflictingDevices [ Line3 ] Values { CaptureCTL "_ucm0001.hw:M2" CaptureChannels 1 CapturePCM "_ucm0001.minifuse12_mono_in:M2,0,1" CapturePriority 300 PlaybackCTL "_ucm0001.hw:M2" } } } Verb.Direct { Comment "Direct MiniFuse 2" Device.Direct { Comment "Direct MiniFuse 2" Values { CaptureCTL "_ucm0001.hw:M2" CaptureChannels 4 CapturePCM "_ucm0001.hw:M2" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:M2" PlaybackChannels 4 PlaybackPCM "_ucm0001.hw:M2" PlaybackPriority 1000 } } } ``` Pipewire debug log "spa.alsa:D" excerpts (I have no idea how to get this elsewhere, but it matches to offered reality): ``` [I][12125.973616] spa.alsa | [ alsa-pcm.c: 618 spa_alsa_open()] 0x562ee6e95448: ALSA device open '_ucm0001.minifuse12_mono_in:M2,0,0' capture [I][12125.993003] spa.alsa | [ alsa-pcm.c: 568 probe_pitch_ctl()] minifuse12_mono_in:M2,0,0 could not find ctl device: No such file or directory [D][12125.993047] spa.alsa | [ alsa-pcm.c: 1048 debug_hw_params()] enum_pcm_formats: [D][12125.993065] spa.alsa | [ alsa-pcm.c: 469 log_write()] ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED [D][12125.993075] spa.alsa | [ alsa-pcm.c: 469 log_write()] FORMAT: S32_LE [D][12125.993082] spa.alsa | [ alsa-pcm.c: 469 log_write()] SUBFORMAT: STD [D][12125.993089] spa.alsa | [ alsa-pcm.c: 469 log_write()] SAMPLE_BITS: 32 [D][12125.993095] spa.alsa | [ alsa-pcm.c: 469 log_write()] FRAME_BITS: 32 [D][12125.993101] spa.alsa | [ alsa-pcm.c: 469 log_write()] CHANNELS: 1 [D][12125.993107] spa.alsa | [ alsa-pcm.c: 469 log_write()] RATE: 48000 [D][12125.993113] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_TIME: 10000 [D][12125.993119] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_SIZE: 480 [D][12125.993124] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_BYTES: 1920 [D][12125.993130] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIODS: [2 50] [D][12125.993136] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_TIME: [20000 500000] [D][12125.993142] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_SIZE: [960 24000] [D][12125.993148] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_BYTES: [3840 96000] [D][12125.993153] spa.alsa | [ alsa-pcm.c: 469 log_write()] TICK_TIME: ALL [D][12125.993165] spa.alsa | [ alsa-pcm.c: 885 add_rate()] min:48000 max:48000 min-allowed:0 scale:1 interleave:1 all:0 [D][12125.993172] spa.alsa | [ alsa-pcm.c: 909 add_rate()] rate:48000 multi:1 card:0 def:0 [D][12125.993180] spa.alsa | [ alsa-pcm.c: 963 add_channels()] channels (1 1) default:1 all:0 [D][12125.993187] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x562ee6e95448: position 0 2 ``` ``` [I][12125.637937] spa.alsa | [ alsa-pcm.c: 618 spa_alsa_open()] 0x562ee6e24218: ALSA device open '_ucm0001.minifuse12_stereo_out:M2,0,0,1' playback [I][12125.657353] spa.alsa | [ alsa-pcm.c: 568 probe_pitch_ctl()] minifuse12_stereo_out:M2,0,0,1 could not find ctl device: No such file or directory [D][12125.657379] spa.alsa | [ alsa-pcm.c: 1048 debug_hw_params()] enum_pcm_formats: [D][12125.657398] spa.alsa | [ alsa-pcm.c: 469 log_write()] ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED [D][12125.657407] spa.alsa | [ alsa-pcm.c: 469 log_write()] FORMAT: S16_LE [D][12125.657414] spa.alsa | [ alsa-pcm.c: 469 log_write()] SUBFORMAT: STD [D][12125.657420] spa.alsa | [ alsa-pcm.c: 469 log_write()] SAMPLE_BITS: 16 [D][12125.657427] spa.alsa | [ alsa-pcm.c: 469 log_write()] FRAME_BITS: 32 [D][12125.657433] spa.alsa | [ alsa-pcm.c: 469 log_write()] CHANNELS: 2 [D][12125.657439] spa.alsa | [ alsa-pcm.c: 469 log_write()] RATE: 48000 [D][12125.657445] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_TIME: 10000 [D][12125.657452] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_SIZE: 480 [D][12125.657458] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_BYTES: 1920 [D][12125.657463] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIODS: [2 50] [D][12125.657468] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_TIME: [20000 500000] [D][12125.657474] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_SIZE: [960 24000] [D][12125.657481] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_BYTES: [3840 96000] [D][12125.657488] spa.alsa | [ alsa-pcm.c: 469 log_write()] TICK_TIME: ALL [D][12125.657499] spa.alsa | [ alsa-pcm.c: 885 add_rate()] min:48000 max:48000 min-allowed:0 scale:1 interleave:1 all:0 [D][12125.657507] spa.alsa | [ alsa-pcm.c: 909 add_rate()] rate:48000 multi:1 card:0 def:0 [D][12125.657514] spa.alsa | [ alsa-pcm.c: 963 add_channels()] channels (2 2) default:2 all:0 [D][12125.657521] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x562ee6e24218: position 0 3 [D][12125.657528] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x562ee6e24218: position 1 4 ``` Without matching UCM profile the in/outs offers the full range (but everything is mapped as "surround" 4-channel): ``` [D][13133.383190] spa.alsa | [ alsa-pcm.c: 1346 spa_alsa_enum_format()] opened:0 format:0 started:0 [I][13133.383199] spa.alsa | [ alsa-pcm.c: 618 spa_alsa_open()] 0x5629c57893f8: ALSA device open 'surround40:4' capture [I][13133.384911] spa.alsa | [ alsa-pcm.c: 568 probe_pitch_ctl()] surround40:4 could not find ctl device: No such file or directory [D][13133.384928] spa.alsa | [ alsa-pcm.c: 1048 debug_hw_params()] enum_pcm_formats: [D][13133.384941] spa.alsa | [ alsa-pcm.c: 469 log_write()] ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED [D][13133.384947] spa.alsa | [ alsa-pcm.c: 469 log_write()] FORMAT: S32_LE [D][13133.384951] spa.alsa | [ alsa-pcm.c: 469 log_write()] SUBFORMAT: STD [D][13133.384955] spa.alsa | [ alsa-pcm.c: 469 log_write()] SAMPLE_BITS: 32 [D][13133.384959] spa.alsa | [ alsa-pcm.c: 469 log_write()] FRAME_BITS: 128 [D][13133.384963] spa.alsa | [ alsa-pcm.c: 469 log_write()] CHANNELS: 4 [D][13133.384967] spa.alsa | [ alsa-pcm.c: 469 log_write()] RATE: [44100 192000] [D][13133.384970] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_TIME: [125 1000000] [D][13133.384974] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_SIZE: [6 192000] [D][13133.384978] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_BYTES: [96 3072000] [D][13133.384982] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIODS: [2 1024] [D][13133.384986] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_TIME: (62 2000000] [D][13133.384990] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_SIZE: [12 384000] [D][13133.384993] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_BYTES: [192 6144000] [D][13133.384997] spa.alsa | [ alsa-pcm.c: 469 log_write()] TICK_TIME: ALL [D][13133.385004] spa.alsa | [ alsa-pcm.c: 885 add_rate()] min:44100 max:192000 min-allowed:0 scale:1 interleave:1 all:0 [D][13133.385009] spa.alsa | [ alsa-pcm.c: 909 add_rate()] rate:48000 multi:1 card:0 def:0 [D][13133.385014] spa.alsa | [ alsa-pcm.c: 963 add_channels()] channels (4 4) default:4 all:0 [D][13133.385018] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57893f8: position 0 3 [D][13133.385022] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57893f8: position 1 4 [D][13133.385026] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57893f8: position 2 12 [D][13133.385030] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57893f8: position 3 13 ``` ``` [D][13133.381023] spa.alsa | [ alsa-pcm.c: 1346 spa_alsa_enum_format()] opened:0 format:0 started:0 [I][13133.381032] spa.alsa | [ alsa-pcm.c: 618 spa_alsa_open()] 0x5629c57650b8: ALSA device open 'surround40:4' playback [I][13133.382756] spa.alsa | [ alsa-pcm.c: 568 probe_pitch_ctl()] surround40:4 could not find ctl device: No such file or directory [D][13133.382772] spa.alsa | [ alsa-pcm.c: 1048 debug_hw_params()] enum_pcm_formats: [D][13133.382786] spa.alsa | [ alsa-pcm.c: 469 log_write()] ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED [D][13133.382792] spa.alsa | [ alsa-pcm.c: 469 log_write()] FORMAT: S16_LE S32_LE [D][13133.382796] spa.alsa | [ alsa-pcm.c: 469 log_write()] SUBFORMAT: STD [D][13133.382800] spa.alsa | [ alsa-pcm.c: 469 log_write()] SAMPLE_BITS: [16 32] [D][13133.382804] spa.alsa | [ alsa-pcm.c: 469 log_write()] FRAME_BITS: [64 128] [D][13133.382808] spa.alsa | [ alsa-pcm.c: 469 log_write()] CHANNELS: 4 [D][13133.382812] spa.alsa | [ alsa-pcm.c: 469 log_write()] RATE: [44100 192000] [D][13133.382816] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_TIME: [125 1000000] [D][13133.382820] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_SIZE: [6 192000] [D][13133.382824] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_BYTES: [64 3072000] [D][13133.382827] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIODS: [2 1024] [D][13133.382831] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_TIME: (62 2000000] [D][13133.382835] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_SIZE: [12 384000] [D][13133.382839] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_BYTES: [96 6144000] [D][13133.382843] spa.alsa | [ alsa-pcm.c: 469 log_write()] TICK_TIME: ALL [D][13133.382850] spa.alsa | [ alsa-pcm.c: 885 add_rate()] min:44100 max:192000 min-allowed:0 scale:1 interleave:1 all:0 [D][13133.382855] spa.alsa | [ alsa-pcm.c: 909 add_rate()] rate:48000 multi:1 card:0 def:0 [D][13133.382859] spa.alsa | [ alsa-pcm.c: 963 add_channels()] channels (4 4) default:4 all:0 [D][13133.382864] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57650b8: position 0 3 [D][13133.382868] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57650b8: position 1 4 [D][13133.382872] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57650b8: position 2 12 [D][13133.382876] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57650b8: position 3 13 ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/333 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Arturia Minifuse 2 (and probably any other using SplitPCM macro) - lost sample rate range on each in/out (and forced to 48000)
by GitHub issues - opened 29 Jun '23

29 Jun '23

alsa-project/alsa-ucm-conf issue #333 was opened from kmarty: Hi, I have Arturia MiniFuse 2 card allowing playback/capture sample rate in range 44100 - 192000. But using channel name (e.g. `Mic/Line/Inst 1 (L)`, `Main Output L/R`) created by UCM, it offers 48000 only (and in case o playback, it offers S15_LE only). Looks like something is lost when `SplitPCM` macro is used or so (I have absolutely no idea what's going on here). Is there any possibility to make it work as it should? ================================= `/proc/asound/card4/stream0` output: ``` ARTURIA MiniFuse 2 at usb-0000:16:00.0-6.2, high speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S32_LE Channels: 4 Endpoint: 0x01 (1 OUT) (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE Sync Endpoint: 0x81 (1 IN) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: No Interface 1 Altset 2 Format: S16_LE Channels: 4 Endpoint: 0x01 (1 OUT) (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 16 Channel map: FL FR FC LFE Sync Endpoint: 0x81 (1 IN) Sync EP Interface: 1 Sync EP Altset: 2 Implicit Feedback Mode: No Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 4 Endpoint: 0x82 (2 IN) (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE ``` `alsaucm -c hw:4 dump text` output: ``` Verb.HiFi { Comment "Default Alsa Profile" Device.Line1 { Comment "Main Output L/R" Values { CaptureCTL "_ucm0001.hw:M2" PlaybackCTL "_ucm0001.hw:M2" PlaybackChannels 2 PlaybackPCM "_ucm0001.minifuse12_stereo_out:M2,0,0,1" PlaybackPriority 300 } } Device.Line2 { Comment "Loopback L/R" Values { CaptureCTL "_ucm0001.hw:M2" CaptureChannels 2 CapturePCM "_ucm0001.minifuse12_stereo_in:M2,0,2,3" CapturePriority 200 PlaybackCTL "_ucm0001.hw:M2" PlaybackChannels 2 PlaybackPCM "_ucm0001.minifuse12_stereo_out:M2,0,2,3" PlaybackPriority 200 } } Device.Line3 { Comment "Stereo Input 1+2 L/R" ConflictingDevices [ Mic2, Mic1 ] Values { CaptureCTL "_ucm0001.hw:M2" CaptureChannels 2 CapturePCM "_ucm0001.minifuse12_stereo_in:M2,0,0,1" CapturePriority 100 PlaybackCTL "_ucm0001.hw:M2" } } Device.Mic1 { Comment "Mic/Line/Inst 1 (L)" ConflictingDevices [ Line3 ] Values { CaptureCTL "_ucm0001.hw:M2" CaptureChannels 1 CapturePCM "_ucm0001.minifuse12_mono_in:M2,0,0" CapturePriority 400 PlaybackCTL "_ucm0001.hw:M2" } } Device.Mic2 { Comment "Mic/Line/Inst 2 (R)" ConflictingDevices [ Line3 ] Values { CaptureCTL "_ucm0001.hw:M2" CaptureChannels 1 CapturePCM "_ucm0001.minifuse12_mono_in:M2,0,1" CapturePriority 300 PlaybackCTL "_ucm0001.hw:M2" } } } Verb.Direct { Comment "Direct MiniFuse 2" Device.Direct { Comment "Direct MiniFuse 2" Values { CaptureCTL "_ucm0001.hw:M2" CaptureChannels 4 CapturePCM "_ucm0001.hw:M2" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:M2" PlaybackChannels 4 PlaybackPCM "_ucm0001.hw:M2" PlaybackPriority 1000 } } } ``` Pipewire debug log "spa.alsa:D" excerpts (I have no idea how to get this elsewhere, but it matches to offered reality): ``` [I][12125.973616] spa.alsa | [ alsa-pcm.c: 618 spa_alsa_open()] 0x562ee6e95448: ALSA device open '_ucm0001.minifuse12_mono_in:M2,0,0' capture [I][12125.993003] spa.alsa | [ alsa-pcm.c: 568 probe_pitch_ctl()] minifuse12_mono_in:M2,0,0 could not find ctl device: No such file or directory [D][12125.993047] spa.alsa | [ alsa-pcm.c: 1048 debug_hw_params()] enum_pcm_formats: [D][12125.993065] spa.alsa | [ alsa-pcm.c: 469 log_write()] ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED [D][12125.993075] spa.alsa | [ alsa-pcm.c: 469 log_write()] FORMAT: S32_LE [D][12125.993082] spa.alsa | [ alsa-pcm.c: 469 log_write()] SUBFORMAT: STD [D][12125.993089] spa.alsa | [ alsa-pcm.c: 469 log_write()] SAMPLE_BITS: 32 [D][12125.993095] spa.alsa | [ alsa-pcm.c: 469 log_write()] FRAME_BITS: 32 [D][12125.993101] spa.alsa | [ alsa-pcm.c: 469 log_write()] CHANNELS: 1 [D][12125.993107] spa.alsa | [ alsa-pcm.c: 469 log_write()] RATE: 48000 [D][12125.993113] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_TIME: 10000 [D][12125.993119] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_SIZE: 480 [D][12125.993124] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_BYTES: 1920 [D][12125.993130] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIODS: [2 50] [D][12125.993136] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_TIME: [20000 500000] [D][12125.993142] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_SIZE: [960 24000] [D][12125.993148] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_BYTES: [3840 96000] [D][12125.993153] spa.alsa | [ alsa-pcm.c: 469 log_write()] TICK_TIME: ALL [D][12125.993165] spa.alsa | [ alsa-pcm.c: 885 add_rate()] min:48000 max:48000 min-allowed:0 scale:1 interleave:1 all:0 [D][12125.993172] spa.alsa | [ alsa-pcm.c: 909 add_rate()] rate:48000 multi:1 card:0 def:0 [D][12125.993180] spa.alsa | [ alsa-pcm.c: 963 add_channels()] channels (1 1) default:1 all:0 [D][12125.993187] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x562ee6e95448: position 0 2 ``` ``` [I][12125.637937] spa.alsa | [ alsa-pcm.c: 618 spa_alsa_open()] 0x562ee6e24218: ALSA device open '_ucm0001.minifuse12_stereo_out:M2,0,0,1' playback [I][12125.657353] spa.alsa | [ alsa-pcm.c: 568 probe_pitch_ctl()] minifuse12_stereo_out:M2,0,0,1 could not find ctl device: No such file or directory [D][12125.657379] spa.alsa | [ alsa-pcm.c: 1048 debug_hw_params()] enum_pcm_formats: [D][12125.657398] spa.alsa | [ alsa-pcm.c: 469 log_write()] ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED [D][12125.657407] spa.alsa | [ alsa-pcm.c: 469 log_write()] FORMAT: S16_LE [D][12125.657414] spa.alsa | [ alsa-pcm.c: 469 log_write()] SUBFORMAT: STD [D][12125.657420] spa.alsa | [ alsa-pcm.c: 469 log_write()] SAMPLE_BITS: 16 [D][12125.657427] spa.alsa | [ alsa-pcm.c: 469 log_write()] FRAME_BITS: 32 [D][12125.657433] spa.alsa | [ alsa-pcm.c: 469 log_write()] CHANNELS: 2 [D][12125.657439] spa.alsa | [ alsa-pcm.c: 469 log_write()] RATE: 48000 [D][12125.657445] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_TIME: 10000 [D][12125.657452] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_SIZE: 480 [D][12125.657458] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_BYTES: 1920 [D][12125.657463] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIODS: [2 50] [D][12125.657468] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_TIME: [20000 500000] [D][12125.657474] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_SIZE: [960 24000] [D][12125.657481] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_BYTES: [3840 96000] [D][12125.657488] spa.alsa | [ alsa-pcm.c: 469 log_write()] TICK_TIME: ALL [D][12125.657499] spa.alsa | [ alsa-pcm.c: 885 add_rate()] min:48000 max:48000 min-allowed:0 scale:1 interleave:1 all:0 [D][12125.657507] spa.alsa | [ alsa-pcm.c: 909 add_rate()] rate:48000 multi:1 card:0 def:0 [D][12125.657514] spa.alsa | [ alsa-pcm.c: 963 add_channels()] channels (2 2) default:2 all:0 [D][12125.657521] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x562ee6e24218: position 0 3 [D][12125.657528] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x562ee6e24218: position 1 4 ``` Without matching UCM profile the in/outs offers the full range (but everything is mapped as "surround" 4-channel): ``` [D][13133.383190] spa.alsa | [ alsa-pcm.c: 1346 spa_alsa_enum_format()] opened:0 format:0 started:0 [I][13133.383199] spa.alsa | [ alsa-pcm.c: 618 spa_alsa_open()] 0x5629c57893f8: ALSA device open 'surround40:4' capture [I][13133.384911] spa.alsa | [ alsa-pcm.c: 568 probe_pitch_ctl()] surround40:4 could not find ctl device: No such file or directory [D][13133.384928] spa.alsa | [ alsa-pcm.c: 1048 debug_hw_params()] enum_pcm_formats: [D][13133.384941] spa.alsa | [ alsa-pcm.c: 469 log_write()] ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED [D][13133.384947] spa.alsa | [ alsa-pcm.c: 469 log_write()] FORMAT: S32_LE [D][13133.384951] spa.alsa | [ alsa-pcm.c: 469 log_write()] SUBFORMAT: STD [D][13133.384955] spa.alsa | [ alsa-pcm.c: 469 log_write()] SAMPLE_BITS: 32 [D][13133.384959] spa.alsa | [ alsa-pcm.c: 469 log_write()] FRAME_BITS: 128 [D][13133.384963] spa.alsa | [ alsa-pcm.c: 469 log_write()] CHANNELS: 4 [D][13133.384967] spa.alsa | [ alsa-pcm.c: 469 log_write()] RATE: [44100 192000] [D][13133.384970] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_TIME: [125 1000000] [D][13133.384974] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_SIZE: [6 192000] [D][13133.384978] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_BYTES: [96 3072000] [D][13133.384982] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIODS: [2 1024] [D][13133.384986] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_TIME: (62 2000000] [D][13133.384990] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_SIZE: [12 384000] [D][13133.384993] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_BYTES: [192 6144000] [D][13133.384997] spa.alsa | [ alsa-pcm.c: 469 log_write()] TICK_TIME: ALL [D][13133.385004] spa.alsa | [ alsa-pcm.c: 885 add_rate()] min:44100 max:192000 min-allowed:0 scale:1 interleave:1 all:0 [D][13133.385009] spa.alsa | [ alsa-pcm.c: 909 add_rate()] rate:48000 multi:1 card:0 def:0 [D][13133.385014] spa.alsa | [ alsa-pcm.c: 963 add_channels()] channels (4 4) default:4 all:0 [D][13133.385018] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57893f8: position 0 3 [D][13133.385022] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57893f8: position 1 4 [D][13133.385026] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57893f8: position 2 12 [D][13133.385030] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57893f8: position 3 13 ``` ``` [D][13133.381023] spa.alsa | [ alsa-pcm.c: 1346 spa_alsa_enum_format()] opened:0 format:0 started:0 [I][13133.381032] spa.alsa | [ alsa-pcm.c: 618 spa_alsa_open()] 0x5629c57650b8: ALSA device open 'surround40:4' playback [I][13133.382756] spa.alsa | [ alsa-pcm.c: 568 probe_pitch_ctl()] surround40:4 could not find ctl device: No such file or directory [D][13133.382772] spa.alsa | [ alsa-pcm.c: 1048 debug_hw_params()] enum_pcm_formats: [D][13133.382786] spa.alsa | [ alsa-pcm.c: 469 log_write()] ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED [D][13133.382792] spa.alsa | [ alsa-pcm.c: 469 log_write()] FORMAT: S16_LE S32_LE [D][13133.382796] spa.alsa | [ alsa-pcm.c: 469 log_write()] SUBFORMAT: STD [D][13133.382800] spa.alsa | [ alsa-pcm.c: 469 log_write()] SAMPLE_BITS: [16 32] [D][13133.382804] spa.alsa | [ alsa-pcm.c: 469 log_write()] FRAME_BITS: [64 128] [D][13133.382808] spa.alsa | [ alsa-pcm.c: 469 log_write()] CHANNELS: 4 [D][13133.382812] spa.alsa | [ alsa-pcm.c: 469 log_write()] RATE: [44100 192000] [D][13133.382816] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_TIME: [125 1000000] [D][13133.382820] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_SIZE: [6 192000] [D][13133.382824] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIOD_BYTES: [64 3072000] [D][13133.382827] spa.alsa | [ alsa-pcm.c: 469 log_write()] PERIODS: [2 1024] [D][13133.382831] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_TIME: (62 2000000] [D][13133.382835] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_SIZE: [12 384000] [D][13133.382839] spa.alsa | [ alsa-pcm.c: 469 log_write()] BUFFER_BYTES: [96 6144000] [D][13133.382843] spa.alsa | [ alsa-pcm.c: 469 log_write()] TICK_TIME: ALL [D][13133.382850] spa.alsa | [ alsa-pcm.c: 885 add_rate()] min:44100 max:192000 min-allowed:0 scale:1 interleave:1 all:0 [D][13133.382855] spa.alsa | [ alsa-pcm.c: 909 add_rate()] rate:48000 multi:1 card:0 def:0 [D][13133.382859] spa.alsa | [ alsa-pcm.c: 963 add_channels()] channels (4 4) default:4 all:0 [D][13133.382864] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57650b8: position 0 3 [D][13133.382868] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57650b8: position 1 4 [D][13133.382872] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57650b8: position 2 12 [D][13133.382876] spa.alsa | [ alsa-pcm.c: 1036 add_channels()] 0x5629c57650b8: position 3 13 ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/333 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook G10
by Huang Rui 29 Jun '23

29 Jun '23

In the commit 3e10f6ca76c4 ("ALSA: hda/realtek: Add quirk for HP EliteBook G10 laptops"), speakers are fixed but mute/micmute LEDs does not lit. Need the ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED to get all mute/micmute LEDs working. Signed-off-by: Huang Rui <vowstar(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dabfdecece26..425c87733e79 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9491,7 +9491,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b65, "HP ProBook 455 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b66, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b70, "HP EliteBook 835 G10", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x103c, 0x8b72, "HP EliteBook 845 G10", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8b72, "HP EliteBook 845 G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b74, "HP EliteBook 845W G10", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8b77, "HP ElieBook 865 G10", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), -- 2.41.0

1 0

[PATCH] ASoC: hdmi-codec: fix channel info for compressed formats
by Matthias Reichl 29 Jun '23

29 Jun '23

According to CTA 861 the channel/speaker allocation info in the audio infoframe only applies to uncompressed (PCM) audio streams. The channel count info should indicate the number of channels in the transmitted audio, which usually won't match the number of channels used to transmit the compressed bitstream. Some devices (eg some Sony TVs) will refuse to decode compressed audio if these values are not set correctly. To fix this we can simply set the channel count to 0 (which means "refer to stream header") and set the channel/speaker allocation to 0 as well (which would mean stereo FL/FR for PCM, a safe value all sinks will support) when transmitting compressed audio. Signed-off-by: Matthias Reichl <hias(a)horus.com> --- sound/soc/codecs/hdmi-codec.c | 36 +++++++++++++++++++++++------------ 1 file changed, 24 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6d980fbc42077..d21f69f053422 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -495,31 +495,43 @@ static int hdmi_codec_fill_codec_params(struct snd_soc_dai *dai, struct hdmi_codec_params *hp) { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); - int idx; - - /* Select a channel allocation that matches with ELD and pcm channels */ - idx = hdmi_codec_get_ch_alloc_table_idx(hcp, channels); - if (idx < 0) { - dev_err(dai->dev, "Not able to map channels to speakers (%d)\n", - idx); - hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; - return idx; + int idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; + u8 ca_id = 0; + bool pcm_audio = !(hcp->iec_status[0] & IEC958_AES0_NONAUDIO); + + if (pcm_audio) { + /* Select a channel allocation that matches with ELD and pcm channels */ + idx = hdmi_codec_get_ch_alloc_table_idx(hcp, channels); + + if (idx < 0) { + dev_err(dai->dev, "Not able to map channels to speakers (%d)\n", + idx); + hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; + return idx; + } + + ca_id = hdmi_codec_channel_alloc[idx].ca_id; } memset(hp, 0, sizeof(*hp)); hdmi_audio_infoframe_init(&hp->cea); - hp->cea.channels = channels; + + if (pcm_audio) + hp->cea.channels = channels; + else + hp->cea.channels = 0; + hp->cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; hp->cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; hp->cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; - hp->cea.channel_allocation = hdmi_codec_channel_alloc[idx].ca_id; + hp->cea.channel_allocation = ca_id; hp->sample_width = sample_width; hp->sample_rate = sample_rate; hp->channels = channels; - hcp->chmap_idx = hdmi_codec_channel_alloc[idx].ca_id; + hcp->chmap_idx = idx; return 0; } -- 2.39.2

3 4

[PATCH 1/6] ASoC: amd: ps: add comments for DMA irq bits mapping
by Vijendar Mukunda 29 Jun '23

29 Jun '23

Add comments for DMA stream id and IRQ bit mapping in ACP_EXTERNAL_CNTL & ACP_EXTERNAL_CNTL1 registers for SDW0 and SDW1 manager instances. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/acp63.h | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index e96e6dc9d90f..733a16e23d32 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -116,8 +116,28 @@ #define ACP63_SDW0_DMA_MAX_STREAMS 6 #define ACP63_SDW1_DMA_MAX_STREAMS 2 #define ACP_P1_AUDIO_TX_THRESHOLD 6 + +/* + * Below entries describes SDW0 instance DMA stream id and DMA irq bit mapping + * in ACP_EXTENAL_INTR_CNTL register. + * Stream id IRQ Bit + * 0 (SDW0_AUDIO0_TX) 28 + * 1 (SDW0_AUDIO1_TX) 26 + * 2 (SDW0_AUDIO2_TX) 24 + * 3 (SDW0_AUDIO0_RX) 27 + * 4 (SDW0_AUDIO1_RX) 25 + * 5 (SDW0_AUDIO2_RX) 23 + */ #define SDW0_DMA_TX_IRQ_MASK(i) (ACP_AUDIO0_TX_THRESHOLD - (2 * (i))) #define SDW0_DMA_RX_IRQ_MASK(i) (ACP_AUDIO0_RX_THRESHOLD - (2 * (i))) + +/* + * Below entries describes SDW1 instance DMA stream id and DMA irq bit mapping + * in ACP_EXTENAL_INTR_CNTL1 register. + * Stream id IRQ Bit + * 0 (SDW1_AUDIO1_TX) 6 + * 1 (SDW1_AUDIO1_RX) 5 + */ #define SDW1_DMA_IRQ_MASK(i) (ACP_P1_AUDIO_TX_THRESHOLD - (i)) #define ACP_DELAY_US 5 -- 2.34.1

2 6

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NPx0SNx
by Werner Sembach 29 Jun '23

29 Jun '23

This applies a SND_PCI_QUIRK(...) to the Clevo NPx0SNx barebones fixing the microphone not being detected on the headset combo port. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- 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 f595a250fab33..d2ac56398d554 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9642,6 +9642,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.34.1

2 1

Fix symver build error on non-ELF platforms
by GitHub pull_request - opened 29 Jun '23

29 Jun '23

alsa-project/alsa-lib pull request #334 was opened from bkuhls: The following error is observed on Microblaze [1] build: error: symver is only supported on ELF platforms due to using __attribute__((symver)) on non-ELF platform. [1] http://autobuild.buildroot.net/results/1e9/1e965d83d75615f35308440c5db04431… ac_check_attribute_symver.m4 was downloaded from https://github.com/smuellerDD/libkcapi/blob/master/m4/ac_check_attribute_sy… Request URL : https://github.com/alsa-project/alsa-lib/pull/334 Patch URL : https://github.com/alsa-project/alsa-lib/pull/334.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

MIC Capture Stops due to Playback
by Arjav Parikh 29 Jun '23

29 Jun '23

Hi I am using alsa api's in my C Program to implement two-way talk i.e capturing data from mic and playing sound on speaker received from Cloud. I am using a Microsemi ZL38050 Audio DSP connected to an Ambarella CV25 SOC via I2S/SPI. Randomly I have observed that If I stop and resume receiving data from Cloud to play it on speaker using snd_pcm_writei() then my capture path stops. I do not hear any sound captured from MIC. Can someone help me to debug this issue because when this issue occurs there are no errors reported from Alsa nor do I see anything in dmesg as well. Thanks & Regards, Arjav Parikh

1 0

configure: Add large file support via CFLAGS
by GitHub pull_request - opened 29 Jun '23

29 Jun '23

alsa-project/alsa-lib pull request #333 was opened from kendase3: Currently scanelf (pax-utils) detects missing LFS support in the following files/calls: fopen,__open_2,mmap,lseek,open /usr/lib/libasound.so.2.0.0 fopen,open /usr/lib/libatopology.so.2.0.0 Note that this may cause problems on systems with a 32-bit kernel, but I've tested playing audio on a more recent 32-bit system with a 64-bit kernel. Notably AC_SYS_LARGEFILE (currently used in alsa-utils for example) did not solve the issue for my case, whereas FILE_OFFSET_BITS did. Let me know if you'd prefer an alternative approach! Request URL : https://github.com/alsa-project/alsa-lib/pull/333 Patch URL : https://github.com/alsa-project/alsa-lib/pull/333.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/3] ASoC: qcom: get tplg firmware-name from device tree
by Srinivas Kandagatla 28 Jun '23

28 Jun '23

With the existing setup the firmware file is looked up in a fixed location based on soundcard name, however this is not really scalable and not inline with where linux-firmware stores board specific firmwares. This patchset adds a new binding to q6apm to allow board specific Device Tree to pass firmware-name, this also makes it inline with how other firmwares are specified in qcom boards. Srinivas Kandagatla (3): ASoC: dt-bindings: q6apm: add firmware-name bindings ASoC: qcom: q6apm: add support for reading firmware name from DT arm64: dts: qcom: sc8280xp-x13s: Add audio topology firmware path .../devicetree/bindings/sound/qcom,q6apm.yaml | 5 +++++ .../dts/qcom/sc8280xp-lenovo-thinkpad-x13s.dts | 5 +++++ sound/soc/qcom/qdsp6/topology.c | 14 ++++++-------- 3 files changed, 16 insertions(+), 8 deletions(-) -- 2.21.0

4 15

ucm2: USB-Audio: Add support for ALC4080 Gigabyte Z590 AORUS ULTRA
by GitHub pull_request - reopened 28 Jun '23

28 Jun '23

alsa-project/alsa-ucm-conf pull request #310 was reopened from BearsPunch: When installed Manjaro I had an issue my audio card was not recognized. USB ID is 0414:a00d and as I see you already have it under if.realtek-alc1220-vb condition, but I have ALC4080. So I manually changed USB-Audio.conf under If.realtek-alc4080 condition from (0414:a0(0e|1[0124])) to (0414:a0(0[de]|1[0124])) and now it works perfectly. But Ubuntu (maybe from 22.04) doesn't have this issue. Does Canonical update alsa-ucm-conf by themselves without pull request to main branch? Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/310 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/310.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: ump: Correct wrong byte size at converting a UMP System message
by Takashi Iwai 28 Jun '23

28 Jun '23

A wrong size for UMP_SYSTEM_STATUS_MIDI_TIME_CODE and case UMP_SYSTEM_STATUS_SONG_SELECT was reported at converting to the legacy MIDI 1.0 stream. This patch corrects the value. Fixes: 0b5288f5fe63 ("ALSA: ump: Add legacy raw MIDI support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/ump_convert.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/ump_convert.c b/sound/core/ump_convert.c index fb61df424a87..de04799fdb69 100644 --- a/sound/core/ump_convert.c +++ b/sound/core/ump_convert.c @@ -73,7 +73,7 @@ static int cvt_ump_system_to_legacy(u32 data, unsigned char *buf) case UMP_SYSTEM_STATUS_MIDI_TIME_CODE: case UMP_SYSTEM_STATUS_SONG_SELECT: buf[1] = (data >> 8) & 0x7f; - return 1; + return 2; case UMP_SYSTEM_STATUS_SONG_POSITION: buf[1] = (data >> 8) & 0x7f; buf[2] = data & 0x7f; -- 2.35.3

1 0

[PATCH] ALSA: fireface: make read-only const array for model names static
by Takashi Sakamoto 28 Jun '23

28 Jun '23

It is preferable not to populate the constant array for constant strings on the stack. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c index 82241058ea14..6e84e4787259 100644 --- a/sound/firewire/fireface/ff.c +++ b/sound/firewire/fireface/ff.c @@ -16,7 +16,7 @@ MODULE_LICENSE("GPL"); static void name_card(struct snd_ff *ff) { struct fw_device *fw_dev = fw_parent_device(ff->unit); - const char *const names[] = { + static const char *const names[] = { [SND_FF_UNIT_VERSION_FF800] = "Fireface800", [SND_FF_UNIT_VERSION_FF400] = "Fireface400", [SND_FF_UNIT_VERSION_UFX] = "FirefaceUFX", -- 2.39.2

2 1

[PATCH][next] ALSA: oxfw: make read-only const array models static
by Colin Ian King 28 Jun '23

28 Jun '23

Don't populate the const array on the stack, instead make it static. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/firewire/oxfw/oxfw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 9523479fa94a..63d40f1a914f 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -44,7 +44,7 @@ struct compat_info { static bool detect_loud_models(struct fw_unit *unit) { - const char *const models[] = { + static const char *const models[] = { "Onyxi", "Onyx-i", "Onyx 1640i", -- 2.39.2

3 2

acp62: add initial support for AMD ACP v6.2 RPL
by GitHub pull_request - edited 28 Jun '23

28 Jun '23

alsa-project/alsa-ucm-conf pull request #332 was edited from pseudocc: See also: [config SND_SOC_AMD_RPL_ACP6x](https://github.com/torvalds/linux/commit/4fb35936a37… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/332 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/332.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Qualcomm: sc8280xp: fix device numbers
by GitHub pull_request - opened 28 Jun '23

28 Jun '23

alsa-project/alsa-ucm-conf pull request #331 was opened from Srinivas-Kandagatla: Due to a bug in the Qualcomm ASoC drivers, we ended up with card device numbers starting at some number that is not dai-link id. This bug is now fixed, so update the ucm files inline with this. Fixes: https://github.com/alsa-project/alsa-ucm-conf/pull/284 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/331 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/331.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

topology: pre-processor: Add support for object arrays
by GitHub pull_request - edited 27 Jun '23

27 Jun '23

alsa-project/alsa-utils pull request #216 was edited from ranj063: Introduce a new keyword, "CombineArrays" to instantiate multiple nodes of the type specified. For example: CombineArrays.Object.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } ] This would be expanded into 6 objects with the rate, bit_depth and valid_bit_depth combinations of the arrays of values above. Object.Base.input_audio_format [ { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 24 } ] The CombineArrays definition is an array so that multiple combinations can be specified in order to deal with only valid combinations. For example, 16-bit bit_depth is only valid with 16-bit valid_bit_depth. So if we also want to add those combinations with the multiple rates, the definition would look like: CombineArrays.Object.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } { in_rate [ 8000 16000 48000 ] in_bit_depth [ 16 ] in_valid_bit_depth [ 16 ] } ] Request URL : https://github.com/alsa-project/alsa-utils/pull/216 Patch URL : https://github.com/alsa-project/alsa-utils/pull/216.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Intel/sof-essx8336: Fix HiFi.conf
by GitHub pull_request - edited 27 Jun '23

27 Jun '23

alsa-project/alsa-ucm-conf pull request #328 was edited from xry111: 1. <del>Enable Left/Right Headphone Mixer Left/Right DAC Switch in BootSequence. Despite they seem enabled in sof-essx8336.conf, if I don't add this I'll need to enable them with alsamixer. I guess they are killed by disdevall.</del> 2. Switch Headphone on/off in Headphones enable/disable sequence. This should be obvious. 3. "Headset Switch" is not recognized and triggers errors running "alsaucm -c hw:0 set _verb Hifi". Change it to "Headset Mic Switch". And in the disable sequence we should turn it off, not on. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/328 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/328.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: sb: fix potential deadlock on &chip->mixer_lock
by Chengfeng Ye 27 Jun '23

27 Jun '23

As &chip->mixer_lock is also acquired by the irq snd_sb8dsp_interrupt() which executes under hard-irq context, code executing under process context should disable irq before acquiring the lock, otherwise deadlock could happen if the process context hold the lock then preempt by the interruption. As the ALSA Driver document described, PCM prepare callbacks are not executed with irq disabled by default, thus the acquiring of &chip->mixer_lock should be irq disabled. Possible deadlock scenario: snd_sb8_playback_prepare -> spin_lock(&chip->mixer_lock); <irq interrupt> -> snd_sb8dsp_interrupt() -> snd_sb8_capture_trigger() -> spin_lock(&chip->mixer_lock); (deadlock here) This flaw was found using an experimental static analysis tool we are developing for irq-related deadlock. The tentative patch fix the potential deadlock by spin_lock_irqsave(). Signed-off-by: Chengfeng Ye <dg573847474(a)gmail.com> --- sound/isa/sb/sb8_main.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 2ed176a5a574..81af8dcddcd2 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -148,10 +148,12 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, format); else if (stereo) { /* set playback stereo mode */ - spin_lock(&chip->mixer_lock); + unsigned long flags; + + spin_lock_irqsave(&chip->mixer_lock, flags); mixreg = snd_sbmixer_read(chip, SB_DSP_STEREO_SW); snd_sbmixer_write(chip, SB_DSP_STEREO_SW, mixreg | 0x02); - spin_unlock(&chip->mixer_lock); + spin_unlock_irqrestore(&chip->mixer_lock, flags); /* Soundblaster hardware programming reference guide, 3-23 */ snd_sbdsp_command(chip, SB_DSP_DMA8_EXIT); @@ -164,12 +166,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) } snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (stereo) { + unsigned long flags; snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); - spin_lock(&chip->mixer_lock); + spin_lock_irqsave(&chip->mixer_lock, flags); /* save output filter status and turn it off */ mixreg = snd_sbmixer_read(chip, SB_DSP_PLAYBACK_FILT); snd_sbmixer_write(chip, SB_DSP_PLAYBACK_FILT, mixreg | 0x20); - spin_unlock(&chip->mixer_lock); + spin_unlock_irqrestore(&chip->mixer_lock, flags); /* just use force_mode16 for temporary storate... */ chip->force_mode16 = mixreg; } else { @@ -289,12 +292,13 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (stereo) { + unsigned long flags; snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); - spin_lock(&chip->mixer_lock); + spin_lock_irqsave(&chip->mixer_lock, flags); /* save input filter status and turn it off */ mixreg = snd_sbmixer_read(chip, SB_DSP_CAPTURE_FILT); snd_sbmixer_write(chip, SB_DSP_CAPTURE_FILT, mixreg | 0x20); - spin_unlock(&chip->mixer_lock); + spin_lock_irqsave(&chip->mixer_lock, flags); /* just use force_mode16 for temporary storate... */ chip->force_mode16 = mixreg; } else { -- 2.17.1

3 2

[PATCH 0/8] ASoC: Use the maple tree register cache for RealTek SoundWire CODECs
by Mark Brown 27 Jun '23

27 Jun '23

The RealTek SoundWire CODECs don't support bulk register operations so they gain no advantage from using the rbtree cache, convert them to use the maple tree cache instead - it is more modern. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (8): ASoC: rt700: Use maple tree register cache ASoC: rt711: Use maple tree register cache ASoC: rt712: Use maple tree register cache ASoC: rt715: Use maple tree register cache ASoC: rt722: Use maple tree register cache ASoC: rt1308: Use maple tree register cache ASoC: rt1316: Use maple tree register cache ASoC: rt1318: Use maple tree register cache sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt1318-sdw.c | 2 +- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt711-sdca-sdw.c | 4 ++-- sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt712-sdca-dmic.c | 4 ++-- sound/soc/codecs/rt712-sdca-sdw.c | 4 ++-- sound/soc/codecs/rt715-sdca-sdw.c | 4 ++-- sound/soc/codecs/rt722-sdca-sdw.c | 4 ++-- 10 files changed, 15 insertions(+), 15 deletions(-) --- base-commit: 08e6c4bb17087584261c4aff555d32fc1c620b81 change-id: 20230609-asoc-rt-sdw-maple-fa6714ac4318 Best regards, -- Mark Brown <broonie(a)kernel.org>

3 14

[PATCH AUTOSEL 4.14 5/5] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 27 Jun '23

27 Jun '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index cc8f06638edca..7226c03f15934 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -50,6 +50,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -184,6 +185,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -332,14 +334,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -349,14 +353,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -381,13 +385,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -401,10 +409,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -423,6 +430,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.39.2

1 0

[PATCH AUTOSEL 4.19 5/5] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 27 Jun '23

27 Jun '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 838c3c8b403cb..2ddfd6fed122e 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -50,6 +50,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -184,6 +185,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -332,14 +334,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -349,14 +353,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -381,13 +385,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -401,10 +409,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -423,6 +430,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.39.2

1 0