February 2025 - Alsa-devel - mailman.alsa-project.org

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic) - XLR out: AES3 only [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are S/PDIF via coax - channels 10 and 11 are AES/EBU or S/PDIF via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK - _ADAT OUT_: TOSLINK - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v6.14-rc2
by Mark Brown 14 Feb '25

14 Feb '25

The following changes since commit 2014c95afecee3e76ca4a56956a936e23283f05b: Linux 6.14-rc1 (2025-02-02 15:39:26 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc2 for you to fetch changes up to 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 More fixes and deviec quirks, most of them driver specific including a few SOF robustness fixes. Nothing super remarkable individually. ---------------------------------------------------------------- Bard Liao (2): ASoC: Intel: soc-acpi-intel-tgl-match: declare adr as ull ASoC: Intel: soc-acpi-intel-mtl-match: declare adr as ull Cristian Ciocaltea (4): ASoC: SOF: amd: Add post_fw_run_delay ACP quirk ASoC: SOF: amd: Drop unused includes from Vangogh driver ASoC: SOF: amd: Handle IPC replies before FW_BOOT_COMPLETE ASoC: SOF: amd: Add branch prediction hint in ACP IRQ handler John Keeping (1): ASoC: rockchip: i2s-tdm: fix shift config for SND_SOC_DAIFMT_DSP_[AB] Kuninori Morimoto (4): ASoC: simple-card-utils.c: add missing dlc->of_node ASoC: rsnd: indicate unsupported clock rate ASoC: rsnd: don't indicate warning on rsnd_kctrl_accept_runtime() ASoC: rsnd: adjust convert rate limitation Mark Brown (5): ASoC: Intel: soc-acpi-intel-tgl/mtl-match: declare adr ASoC: rsnd: adjust convert rate in 1% ASoC: SOF: Correct sps->stream and cstream nullity ASoC: Intel: sof_sdw: Add support for new devices ASoC: Intel: soc-acpi-intel-ptl-match typo fixups Nikita Zhandarovich (1): ASoC: fsl_micfil: Enable default case in micfil_set_quality() Peter Ujfalusi (5): ASoC: SOF: stream-ipc: Check for cstream nullity in sof_ipc_msg_data() ASoC: SOF: pcm: Clear the susbstream pointer to NULL on close ASoC: SOF: ipc4-topology: Harden loops for looking up ALH copiers ASoC: Intel: soc-acpi-intel-ptl-match: revise typo of rt712_vb + rt1320 support ASoC: Intel: soc-acpi-intel-ptl-match: revise typo of rt713_vb_l2_rt1320_l13 Richard Fitzgerald (2): ASoC: Intel: sof_sdw: Add lookup of quirk using PCI subsystem ID ASoC: Intel: sof_sdw: Add quirk for Asus Zenbook S14 Shenghao Ding (1): ASoC: tas2781: drop a redundant code Shengjiu Wang (1): ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device Stefan Binding (2): ASoC: cs35l41: Fallback to using HID for system_name if no SUB is available ASoC: cs35l41: Fix acpi_device_hid() not found Terry Cheong (1): ASoC: SOF: Intel: hda: add softdep pre to snd-hda-codec-hdmi module Uday M Bhat (1): ASoC: Intel: sof_sdw: Add support for Fatcat board with BT offload enabled in PTL platform Vitaly Rodionov (1): ASoC: arizona/madera: use fsleep() in up/down DAPM event delays. sound/soc/codecs/arizona.c | 14 +-- sound/soc/codecs/cs35l41.c | 30 ++++-- sound/soc/codecs/madera.c | 10 +- sound/soc/codecs/tas2781-i2c.c | 4 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/fsl/fsl_micfil.c | 2 + sound/soc/fsl/imx-audmix.c | 31 ------ sound/soc/generic/simple-card-utils.c | 1 + sound/soc/intel/boards/sof_sdw.c | 41 ++++++-- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 12 +-- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 16 +-- sound/soc/renesas/rcar/core.c | 14 --- sound/soc/renesas/rcar/rsnd.h | 1 - sound/soc/renesas/rcar/src.c | 116 +++++++++++++++++----- sound/soc/renesas/rcar/ssi.c | 3 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 4 +- sound/soc/sof/amd/acp-ipc.c | 25 +++-- sound/soc/sof/amd/acp.c | 1 + sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-vangogh.c | 2 - sound/soc/sof/amd/vangogh.c | 22 +++- sound/soc/sof/intel/hda-codec.c | 1 + sound/soc/sof/ipc4-topology.c | 12 ++- sound/soc/sof/pcm.c | 2 + sound/soc/sof/stream-ipc.c | 6 +- 26 files changed, 245 insertions(+), 136 deletions(-)

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USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are S/PDIF via coax - channels 10 and 11 are AES/EBU or S/PDIF via XLR - channels 12 to 19 are ADAT (8 channels) or S/PDIF (12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK - _ADAT OUT_: TOSLINK - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are S/PDIF via coax - channels 10 and 11 are AES/EBU or S/PDIF via XLR - channels 12 to 19 are ADAT (8 channels) or S/PDIF (12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK - _ADAT OUT_: TOSLINK - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - opened 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was opened from illdefined: [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH V4 0/6] soundwire: amd: code improvements and new platform support
by Vijendar Mukunda 13 Feb '25

13 Feb '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Changes since v3: - Remove extra white line in amd_sdw_manager_start() function. Changes since v2: - Rebase the patch series after dropping the patch which adds conditional check for SoundWire manager device resume when wake event is asserted. Changes since v1: - Drop white line insertion in the code. - Add a conditional check for SoundWire manager device resume when wake event is asserted. - Add new patch to set ACP_PME_EN during runtime suspend sequence - Split the existing implementation of host wake interrupt mask enablement and setting device power state in to two separate patches and refactor set device power state logic. - Drop PME status clear logic. Vijendar Mukunda (6): soundwire: amd: change the soundwire wake enable/disable sequence soundwire: amd: add debug log for soundwire wake event soundwire: amd: add support for ACP7.0 & ACP7.1 platforms soundwire: amd: set device power state during suspend/resume sequence soundwire: amd: set ACP_PME_EN during runtime suspend sequence soundwire: amd: add soundwire host wake interrupt enable/disable sequence drivers/soundwire/amd_manager.c | 149 +++++++++++++++++++++++++++++- drivers/soundwire/amd_manager.h | 26 ++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 172 insertions(+), 5 deletions(-) -- 2.34.1

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[PATCH] ASoC: amd: ps: use switch statements for acp pci revision id check
by Vijendar Mukunda 13 Feb '25

13 Feb '25

Use switch statements for acp pci revision id check in SoundWire dma irq handling. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 220dca8cba85..2ff8e67c19bd 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -111,16 +111,21 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 stream_id = ACP63_SDW0_AUDIO2_RX; break; } - if (adata->acp_rev >= ACP70_PCI_REV) - adata->acp70_sdw0_dma_intr_stat[stream_id] = 1; - else + switch (adata->acp_rev) { + case ACP63_PCI_REV: adata->acp63_sdw0_dma_intr_stat[stream_id] = 1; - + break; + case ACP70_PCI_REV: + case ACP71_PCI_REV: + adata->acp70_sdw0_dma_intr_stat[stream_id] = 1; + break; + } sdw_dma_irq_flag = 1; } } } - if (adata->acp_rev == ACP63_PCI_REV) { + switch (adata->acp_rev) { + case ACP63_PCI_REV: if (ext_intr_stat1 & ACP63_P1_AUDIO1_RX_THRESHOLD) { writel(ACP63_P1_AUDIO1_RX_THRESHOLD, adata->acp63_base + ACP_EXTERNAL_INTR_STAT1); @@ -133,7 +138,9 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 adata->acp63_sdw1_dma_intr_stat[ACP63_SDW1_AUDIO1_TX] = 1; sdw_dma_irq_flag = 1; } - } else { + break; + case ACP70_PCI_REV: + case ACP71_PCI_REV: if (ext_intr_stat1 & ACP70_P1_SDW_DMA_IRQ_MASK) { for (index = ACP70_P1_AUDIO2_RX_THRESHOLD; index <= ACP70_P1_AUDIO0_TX_THRESHOLD; index++) { @@ -166,6 +173,7 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 } } } + break; } return sdw_dma_irq_flag; } -- 2.34.1

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Re: [PATCH v14 29/53] usb: host: xhci-plat: Set XHCI max interrupters if property is present
by Dennis Babcock 12 Feb '25

12 Feb '25

Hi Dennis Louis Babcock jr

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Add HIFI Config support for QCM6490-IDP and QCS6490-RB3Gen2 Qualcomm boards
by GitHub pull_request - opened 12 Feb '25

12 Feb '25

alsa-project/alsa-ucm-conf pull request #508 was opened from mohsRafi: Add HIFI Config support for QCM6490-IDP and QCS6490-RB3Gen2 Qualcomm boards Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/508 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/508.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Add HIFI config support for QCM6490-IDP, QCS6490-RB3Gen2 Qualcomm boards
by GitHub pull_request - opened 12 Feb '25

12 Feb '25

alsa-project/alsa-ucm-conf pull request #507 was opened from mohsRafi: Add HIFI config support for QCM6490-IDP, QCS6490-RB3Gen2 Qualcomm boards Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/507 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/507.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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RE: [Bug 219743] [REGRESSION, BISECTED] ASUS GA402XY Headset mic stopped working after 6.12.2
by Kailang 12 Feb '25

12 Feb '25

Fixup headset MIC no function.

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[PATCH V2 00/23] ASoC: amd: acp70: add soundwire and acp pdm support
by Vijendar Mukunda 11 Feb '25

11 Feb '25

This patch series includes below changes - Add Audio IO support for ACP7.0 and ACP7.1 platforms for SoundWire IO and ACP PDM controller combination. - Add common driver modules for ACP7.0 and ACP7.1 platforms (acp pci driver, Soundwire dma driver, pdm platform driver). - Add SoundWire generic machine driver changes for legacy stack (No DSP enabled) for ACP7.0 & ACP7.1 platforms. - Add SoundWire machines for ACP7.0 & ACP7.1 platforms. Changes since v1: - Change the code sequence for handling SoundWire wake interrupts. - Add new patch to restore host wake interrupt mask - Add new patch to enable ACP PME during ACP init sequence. - Change ACP PCI driver PM ops code sequence. Vijendar Mukunda (23): ASoC: amd: add register header file for ACP7.0 platform ASoC: amd: acp70: add acp pci driver for ACP7.0 and ACP7.1 platforms ASoC: amd: acp70: add acp init and de-init functions ASoC: amd: acp70: add logic for scanning acp child devices ASoC: amd: acp70: create platform devices for acp child nodes ASoC: amd: acp70: enable driver build for ACP7.0 platform ASoC: amd: acp70: add acp pdm platform driver ASoC: amd: acp70: add acp pdm driver dma ops and dai ops ASoC: amd: acp70: add acp soundwire dma driver ASoC: amd: update ACP7.0 KConfig option description ASoC: amd: acp70: add soundwire dma driver dma ops ASoC: amd: acp70: add acp ip interrupt handler ASoC: amd: acp70: add acp pdm driver pm ops ASoC: amd: acp70: add pm ops support for soundwire dma driver ASoC: amd: acp70: add acp driver pm ops support ASoC: amd: acp70: enable wake capability for acp pci driver ASoC: amd: acp70: add soundwire wake interrupt handling ASoC: amd: acp70: enable soundwire host wake irq mask ASoC: amd: acp70: enable acp pme enable during acp init sequence ASoC: amd: acp70: create a device node for soundwire machine driver ASoC: amd: acp: add machine driver changes for ACP7.0 and ACP7.1 platforms ASoC: amd: acp: add RT711, RT714 & RT1316 support for ACP7.0 platform ASoC: amd: acp: amd-acp70-acpi-match: Add rt722 support include/sound/acp70_chip_offset_byte.h | 484 +++++++++++++ sound/soc/amd/Kconfig | 12 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 + sound/soc/amd/acp/acp-sdw-mach-common.c | 34 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 132 ++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 11 + sound/soc/amd/acp70/Makefile | 9 + sound/soc/amd/acp70/acp70-pdm-dma.c | 463 +++++++++++++ sound/soc/amd/acp70/acp70-sdw-dma.c | 586 ++++++++++++++++ sound/soc/amd/acp70/acp70.h | 288 ++++++++ sound/soc/amd/acp70/pci-acp70.c | 833 +++++++++++++++++++++++ sound/soc/amd/mach-config.h | 1 + 14 files changed, 2871 insertions(+), 1 deletion(-) create mode 100644 include/sound/acp70_chip_offset_byte.h create mode 100644 sound/soc/amd/acp/amd-acp70-acpi-match.c create mode 100644 sound/soc/amd/acp70/Makefile create mode 100644 sound/soc/amd/acp70/acp70-pdm-dma.c create mode 100644 sound/soc/amd/acp70/acp70-sdw-dma.c create mode 100644 sound/soc/amd/acp70/acp70.h create mode 100644 sound/soc/amd/acp70/pci-acp70.c -- 2.34.1

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[PATCH V3 00/25] ASoC: amd: Add support for ACP7.0 & ACP7.1 platforms
by Vijendar Mukunda 10 Feb '25

10 Feb '25

This patch series includes the below changes - Refactor existing ACP6.3 platform ACP PCI driver, SoundWire DMA driver code. - Add Audio IO support for ACP7.0 and ACP7.1 platforms for SoundWire IO and ACP PDM controller combination. - Add SoundWire generic machine driver changes for legacy stack (No DSP enabled) for ACP7.0 & ACP7.1 platforms. - Add SoundWire machines for ACP7.0 & ACP7.1 platforms. Changes since v2: - As per Mark's comment, to avoid code duplication drop separate folder(acp70) for ACP7.0 & ACP7.1 platforms and refactor existing ACP6.3 platform ACP PCI driver, SoundWire DMA driver code and add support for ACP7.0 & ACP7.1 platforms. - Add ACP PCI driver hw_ops for ACP6.3, ACP7.0 & ACP7.1 platforms. - Implement platform specific code in Soundwire DMA driver by adding acp pci revision id checks for ACP6.3/ACP7.0/ACP7.1 platforms. - Refactor existing ACP PCI driver Interrupt handler code. - Add new Soundwire machine for RT1320 Speaker + RT722 multi function codec combination. - Add machine driver quirks for Dell SKU's. Changes since v1: - Change the code sequence for handling SoundWire wake interrupts. - Add new patch to restore host wake interrupt mask - Add new patch to enable ACP PME during ACP init sequence. - Change ACP PCI driver PM ops code sequence. Vijendar Mukunda (25): ASoC: amd: ps: rename structure names, variable and other macros ASoC: amd: ps: use macro for ACP6.3 pci revision id ASoC: amd: ps: add acp pci driver hw_ops for acp6.3 platform ASoC: amd: ps: add callback functions for acp pci driver pm ops ASoC: amd: ps: add callback to read acp pin configuration ASoC: amd: ps: add soundwire dma irq thread callback ASoC: amd: ps: refactor soundwire dma interrupt handling ASoC: amd: ps: store acp revision id in SoundWire dma driver private data ASoC: amd: ps: refactor soundwire dma driver code ASoC: amd: ps: refactor soundwire dma interrupts enable/disable sequence ASoC: amd: ps: rename acp_restore_sdw_dma_config() function ASoC: amd: ps: add pci driver hw_ops for ACP7.0 & ACP7.1 variants ASoC: amd: ps: add pm ops related hw_ops for ACP7.0 & ACP7.1 platforms ASoC: amd: ps: add ACP7.0 & ACP7.1 specific soundwire dma driver changes ASoC: amd: ps: implement function to restore dma config for ACP7.0 platform ASoC: amd: ps: add soundwire dma interrupts handling for ACP7.0 platform ASoC: amd: ps: add soundwire wake interrupt handling ASoC: amd: ps: update module description ASoC: amd: ps: update file description and copyright year ASoC: amd: update Pink Sardine platform Kconfig description ASoC: amd: acp: add machine driver changes for ACP7.0 and ACP7.1 platforms ASoC: amd: acp: add RT711, RT714 & RT1316 support for ACP7.0 platform ASoC: amd: acp: amd-acp70-acpi-match: Add rt722 support ASoC: amd: acp: amd-acp70-acpi-match: Add RT1320 & RT722 combination soundwire machine ASoC: amd: amd_sdw: Add quirks for Dell SKU's sound/soc/amd/Kconfig | 10 +- sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 45 +++ sound/soc/amd/acp/acp-sdw-mach-common.c | 34 ++ sound/soc/amd/acp/amd-acp70-acpi-match.c | 160 ++++++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 12 + sound/soc/amd/mach-config.h | 1 + sound/soc/amd/ps/Makefile | 2 +- sound/soc/amd/ps/acp63.h | 246 ++++++++++-- sound/soc/amd/ps/pci-ps.c | 422 +++++++++----------- sound/soc/amd/ps/ps-common.c | 475 +++++++++++++++++++++++ sound/soc/amd/ps/ps-pdm-dma.c | 6 +- sound/soc/amd/ps/ps-sdw-dma.c | 375 ++++++++++++++---- 14 files changed, 1440 insertions(+), 351 deletions(-) create mode 100644 sound/soc/amd/acp/amd-acp70-acpi-match.c create mode 100644 sound/soc/amd/ps/ps-common.c -- 2.34.1

3 28

[PATCH] ALSA: hda/conexant: Add quirk for HP ProBook 450 G4 mute LED
by John Veness 10 Feb '25

10 Feb '25

Allows the LED on the dedicated mute button on the HP ProBook 450 G4 laptop to change colour correctly. Signed-off-by: John Veness <john-linux(a)pelago.org.uk> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 538c37a78..84ab357b8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1080,6 +1080,7 @@ static const struct hda_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x8231, "HP ProBook 450 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), -- 2.48.1

1 0

alsaloop doesn't pass sound with libsamplerate resampling on raspberry pi 5
by GitHub issues - edited 10 Feb '25

10 Feb '25

alsa-project/alsa-utils issue #291 was edited from willangley: Hi alsa-utils team! I'm using `alsaloop`s on a Raspberry Pi 5 as part of a slightly ridiculous signal chain: * Yamaha P-525 keyboard * USB C-to-A cable * Raspberry Pi 5 * USB A-to-B cable * ESI U24XL * SPDIF in/out * Focusrite Clarett+ 4Pre * ... * MacBook Pro [Why? The analog Aux In/Aux Out on the keyboard have an always on loopback connection, which is undesirable when playing to a video meeting. The USB port has user controllable loopback.] Two alsaloop processes run on the Pi, one moving sound from the computer to the piano, one moving sound from the piano to the computer. Both run under `supervisord`. I've got them running at acceptable latency — RTL Utility says 40ms from my MacBook Pro. But I haven't been able to get any of the high quality resampling options working -- everything other than add/drop stops passing sound within a few seconds. I'm surprised; this Pi should be powerful enough to handle this. Any ideas how to troubleshoot it? I didn't see errors in logs at the default log level. Thank you! Will Issue URL : https://github.com/alsa-project/alsa-utils/issues/291 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsaloop doesn't pass sound with libsamplerate resampling on raspberry pi 5
by GitHub issues - edited 10 Feb '25

10 Feb '25

alsa-project/alsa-utils issue #291 was edited from willangley: Hi alsa-utils team! I'm using `alsaloop`s on a Raspberry Pi 5 as part of a slightly ridiculous signal chain: * Yamaha P-525 keyboard * USB C-to-A cable * Raspberry Pi 5 * USB A-to-B cable * ESI U24XL * SPDIF in/out * Focusrite Clarett+ 4Pre * ... * MacBook Pro [Why? The analog Aux In/Aux Out on the keyboard have an always on loopback connection, which is undesirable when playing to a video meeting. The USB port has user controllable loopback.] Two alsaloop processes run on the Pi, one moving sound from the computer to the piano, one moving sound from the piano to the computer. Both run under `supervisord`. I've got them running at acceptable latency — RTL Utility says 40ms from my MacBook Pro. But I haven't been able to get any of the high quality resampling options working -- everything other than add/drop stops passing sound within a few seconds. I'm surprised; this Pi should be powerful enough to handle this. Any ideas how to troubleshoot it? Thank you! Will Issue URL : https://github.com/alsa-project/alsa-utils/issues/291 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsaloop doesn't pass sound with libsamplerate on raspberry pi 5
by GitHub issues - opened 10 Feb '25

10 Feb '25

alsa-project/alsa-utils issue #291 was opened from willangley: Hi alsa-utils team! I'm using `alsaloop`s on a Raspberry Pi 5 as part of a slightly ridiculous signal chain: * Yamaha P-525 keyboard * USB C-to-A cable * Raspberry Pi 5 * USB A-to-B cable * ESI U24XL * SPDIF in/out * Focusrite Clarett+ 4Pre * ... * MacBook Pro [Why? The analog Aux In/Aux Out on the keyboard have an always on loopback connection, which is undesirable when playing to a video meeting. The USB port has user controllable loopback.] Two alsaloop processes run on the Pi, one moving sound from the computer to the piano, one moving sound from the piano to the computer. Both run under `supervisord`. I've got them running at acceptable latency — RTL Utility says 40ms from my MacBook Pro. But I haven't been able to get any of the high quality resampling options working -- everything other than add/drop stops passing sound within a few seconds. I'm surprised; this Pi should be powerful enough to handle this. Any ideas how to troubleshoot it? Thank you! Will Issue URL : https://github.com/alsa-project/alsa-utils/issues/291 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Soundblaster G8 USB Audio not working correctly
by GitHub issues - opened 07 Feb '25

07 Feb '25

alsa-project/alsa-lib issue #437 was opened from Starfoxfs: Soundblaster G8 works partly under Linux Mint 22.1 Cinnamon. Audio playback works in Analog stereo or Duplex mode with headphones and these are displayed as Speakers. Digital SPDIF Out works and also the switch button between headphones and Digital out to external Speakers (only with pipewire not with Pulseaudio). What doesn't work is the Direct Mode and all the Crystalizer surround settings as well as output in virtual 5.1 or 7.1 sound. Microphone doesn´t work, if it plugged in i have no sound at all. [alsa-info.txt](https://github.com/user-attachments/files/18710112/alsa-info… Issue URL : https://github.com/alsa-project/alsa-lib/issues/437 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

rt722: change output volume of speaker to 0dB
by GitHub pull_request - opened 07 Feb '25

07 Feb '25

alsa-project/alsa-ucm-conf pull request #506 was opened from shumingfan: This patch also changes hardware gain to software gain when using volume bar. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/506 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/506.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

soundwire : LnK SoundWire Protocol Analyzer on Linux
by Muni Sekhar 06 Feb '25

06 Feb '25

Hi all, I hope this message finds you well. I am planning to use the LnK SoundWire Protocol Analyzer on a Linux platform. However, I am uncertain whether the current Linux kernel includes the necessary SoundWire drivers to support this . I would appreciate any guidance on how to use the LnK SoundWire Protocol Analyzer with the Linux kernel, as well as information on the specific device drivers required. Could you please provide details on: The compatibility of the LnK SoundWire Protocol Analyzer with the Linux kernel. The steps to set up and use the analyzer on a Linux platform. The necessary device drivers and any additional configurations needed. Thank you for your assistance. -- Thanks, Sekhar

1 0

[PATCH v1] ASoC: tas2781: drop a redundant code
by Shenghao Ding 06 Feb '25

06 Feb '25

Report from internal ticket, priv->cali_data.data devm_kzalloc twice, drop the first one, it is the unnecessary one. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - update the year from 2024 to 2025. - drop a redundant code. --- sound/soc/codecs/tas2781-i2c.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index a730ab6ad4e3..90c5b2e74d12 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -1260,8 +1260,6 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) (cali_data->cali_dat_sz_per_dev + 1) + 1 + 15 + 1; priv->cali_data.total_sz = priv->ndev * (cali_data->cali_dat_sz_per_dev + 1); - priv->cali_data.data = devm_kzalloc(priv->dev, - ext_cali_data->max, GFP_KERNEL); cali_ctrls[i].name = cali_name; cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; cali_ctrls[i].info = snd_soc_bytes_info_ext; -- 2.34.1

2 1

[PATCH v1] MAINTAINERS: remove Kevin Lu from TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS
by Shenghao Ding 04 Feb '25

04 Feb '25

Due to internal re-org, Kevin Lu has no longer maintained audio driver. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 896a307fa065..bc0bf402818d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23370,7 +23370,6 @@ F: sound/soc/ti/ TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org S: Maintained -- 2.34.1

1 0

[PATCH alsa-utils] amixer: implement handling of TLV bytes kcontrols
by Jan Dakinevich 03 Feb '25

03 Feb '25

SND_SOC_BYTES_EXT kcontrols are deprecated and SND_SOC_BYTES_TLV should be used instead. However, there is no support of these kcontrols in amixer. This commit introduces a mechanism to show and write the content of SND_SOC_BYTES_TLV-based kcontrols. Example: $ amixer cset name="Test" 0x10,0x20,0x30,0x40,0x50,0x60,0x70,0x80,0x90,0xa0 $ amixer cget name="Test" numid=42,iface=MIXER,name='Test' ; type=BYTES,access=-----RW-,values=10 : values=0x10,0x20,0x30,0x40,0x50,0x60,0x70,0x80,0x90,0xa0 Signed-off-by: Jan Dakinevich <jan.dakinevich(a)salutedevices.com> --- amixer/amixer.c | 180 +++++++++++++++++++++++++++++++++++++----------- 1 file changed, 141 insertions(+), 39 deletions(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index 07e9819e5cdb..09f774fe2885 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -609,6 +609,8 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, { int err; unsigned int item, idx, count, *tlv; + uint8_t *tlv_bytes = NULL; + unsigned int tlv_bytes_count = 0; snd_ctl_elem_type_t type; snd_ctl_elem_id_t *id; snd_ctl_elem_info_t *info; @@ -628,6 +630,18 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, } count = snd_ctl_elem_info_get_count(info); type = snd_ctl_elem_info_get_type(info); + if (type == SND_CTL_ELEM_TYPE_BYTES && + !snd_ctl_elem_info_is_readable(info) && + !snd_ctl_elem_info_is_writable(info)) { + if (count <= 2 * sizeof(unsigned int)) + return -EINVAL; + tlv_bytes_count = count; + count -= 2 * sizeof(unsigned int); + + tlv_bytes = malloc(tlv_bytes_count); + if (!tlv_bytes) + return -ENOMEM; + } printf("%s; type=%s,access=%s,values=%u", space, control_type(info), control_access(info), count); switch (type) { case SND_CTL_ELEM_TYPE_INTEGER: @@ -661,16 +675,30 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, break; } if (level & LEVEL_BASIC) { - if (!snd_ctl_elem_info_is_readable(info)) + if (snd_ctl_elem_info_is_readable(info)) { + err = snd_hctl_elem_read(elem, control); + if (err < 0) { + error("Control %s element read error: %s\n", card, snd_strerror(err)); + return err; + } + } else if (tlv_bytes) { + err = snd_hctl_elem_tlv_read(elem, (void *)tlv_bytes, tlv_bytes_count); + if (err < 0) { + error("Control %s element TLV read error: %s\n", card, snd_strerror(err)); + free(tlv_bytes); + return err; + } + } else goto __skip_read; - if ((err = snd_hctl_elem_read(elem, control)) < 0) { - error("Control %s element read error: %s\n", card, snd_strerror(err)); - return err; - } + printf("%s: values=", space); for (idx = 0; idx < count; idx++) { if (idx > 0) printf(","); + if (tlv_bytes) { + printf("0x%02x", tlv_bytes[idx + 2 * sizeof(unsigned int)]); + continue; + } switch (type) { case SND_CTL_ELEM_TYPE_BOOLEAN: printf("%s", snd_ctl_elem_value_get_boolean(control, idx) ? "on" : "off"); @@ -699,17 +727,11 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, } } printf("\n"); + if (tlv_bytes) + goto __skip_tlv; __skip_read: if (!snd_ctl_elem_info_is_tlv_readable(info)) goto __skip_tlv; - /* skip ASoC ext bytes controls that may have huge binary TLV data */ - if (type == SND_CTL_ELEM_TYPE_BYTES && - !snd_ctl_elem_info_is_readable(info) && - !snd_ctl_elem_info_is_writable(info)) { - printf("%s; ASoC TLV Byte control, skipping bytes dump\n", space); - goto __skip_tlv; - } - tlv = malloc(4096); if ((err = snd_hctl_elem_tlv_read(elem, tlv, 4096)) < 0) { error("Control %s element TLV read error: %s\n", card, snd_strerror(err)); @@ -720,6 +742,7 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, free(tlv); } __skip_tlv: + free(tlv_bytes); return 0; } @@ -1133,16 +1156,110 @@ static int parse_simple_id(const char *str, snd_mixer_selem_id_t *sid) return 0; } +static int parse_bytes(const char *str, void *ptr, unsigned int len) +{ + char *p; + unsigned long tmp; + unsigned int idx = 0; + unsigned char *data = ptr; + + for (;;) { + if (!*str) + break; + if (idx == len) + return -EINVAL; + + tmp = strtoul(str, &p, 0); + if (p - str < 1) + return -EINVAL; + if (*p == ',') + str = p + 1; + else if (*p == '\0') + str = p; + else + return -EINVAL; + + if (tmp > 255) + return -EINVAL; + + data[idx] = tmp; + idx++; + } + + if (!idx) + return -EINVAL; + + return idx; +} + +static int cset_tlv(int argc, char *argv[], snd_ctl_t *handle, + snd_ctl_elem_info_t *info, snd_ctl_elem_id_t *id) +{ + int err; + unsigned int count; + unsigned int *tlv = NULL; + + count = snd_ctl_elem_info_get_count(info); + if (count <= 2 * sizeof(unsigned int)) { + err = -EINVAL; + goto out; + } + tlv = malloc(count); + if (!tlv) { + err = -ENOMEM; + goto out; + } + err = parse_bytes(argv[1], tlv + 2, + count - 2 * sizeof(unsigned int)); + if (err < 0) + goto out; + tlv[0] = 0; + tlv[1] = err; + + err = snd_ctl_elem_tlv_write(handle, id, tlv); + +out: + free(tlv); + return err; +} + +static int cset_elem(int argc, char *argv[], snd_ctl_t *handle, + snd_ctl_elem_info_t *info, snd_ctl_elem_id_t *id) +{ + int err; + snd_ctl_elem_value_t *control; + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_value_set_id(control, id); + if ((err = snd_ctl_elem_read(handle, control)) < 0) { + if (!ignore_error) + error("Cannot read the given element from control %s\n", card); + return err; + } + err = snd_ctl_ascii_value_parse(handle, control, info, argv[1]); + if (err < 0) { + if (!ignore_error) + error("Control %s parse error: %s\n", card, snd_strerror(err)); + return err; + } + if ((err = snd_ctl_elem_write(handle, control)) < 0) { + if (!ignore_error) + error("Control %s element write error: %s\n", card, snd_strerror(err)); + return err; + } + + return 0; +} + static int cset(int argc, char *argv[], int roflag, int keep_handle) { int err; static snd_ctl_t *handle = NULL; snd_ctl_elem_info_t *info; snd_ctl_elem_id_t *id; - snd_ctl_elem_value_t *control; + snd_ctl_elem_type_t type; snd_ctl_elem_info_alloca(&info); snd_ctl_elem_id_alloca(&id); - snd_ctl_elem_value_alloca(&control); if (argc < 1) { fprintf(stderr, "Specify a full control identifier: [[iface=<iface>,][name='name',][index=<index>,][device=<device>,][subdevice=<subdevice>]]|[numid=<numid>]\n"); @@ -1175,30 +1292,15 @@ static int cset(int argc, char *argv[], int roflag, int keep_handle) } snd_ctl_elem_info_get_id(info, id); /* FIXME: Remove it when hctl find works ok !!! */ if (!roflag) { - snd_ctl_elem_value_set_id(control, id); - if ((err = snd_ctl_elem_read(handle, control)) < 0) { - if (ignore_error) - return 0; - error("Cannot read the given element from control %s\n", card); - if (! keep_handle) { - snd_ctl_close(handle); - handle = NULL; - } - return err; - } - err = snd_ctl_ascii_value_parse(handle, control, info, argv[1]); - if (err < 0) { - if (!ignore_error) - error("Control %s parse error: %s\n", card, snd_strerror(err)); - if (!keep_handle) { - snd_ctl_close(handle); - handle = NULL; - } - return ignore_error ? 0 : err; - } - if ((err = snd_ctl_elem_write(handle, control)) < 0) { - if (!ignore_error) - error("Control %s element write error: %s\n", card, snd_strerror(err)); + type = snd_ctl_elem_info_get_type(info); + if (type == SND_CTL_ELEM_TYPE_BYTES && + !snd_ctl_elem_info_is_readable(info) && + !snd_ctl_elem_info_is_writable(info)) { + err = cset_tlv(argc, argv, handle, info, id); + } else + err = cset_elem(argc, argv, handle, info, id); + + if (err) { if (!keep_handle) { snd_ctl_close(handle); handle = NULL; -- 2.34.1

1 0

Compilation failure with Clang-21 due to macro conflicts
by GitHub issues - opened 02 Feb '25

02 Feb '25

alsa-project/alsa-lib issue #436 was opened from ms178: **Description:** Attempting to compile alsa-lib (both the `git` version and version 1.2.13) with Clang-21 (commit `bfa7edcc6652bdb37d53e0cec64926aab3f280eb`) results in multiple compilation errors. These errors stem from conflicts between macros defined in `midifile.h` and member names within structures defined in `include/alsa/ump_msg.h`. **Environment:** * **alsa-lib version:** `git` (and 1.2.13) * **Compiler:** Clang-21 (`bfa7edcc6652bdb37d53e0cec64926aab3f280eb`) * **Operating System:** CachyOS * **Build System:** Autotools **Steps to Reproduce:** 1. Obtain the PKGBUILD and the other files from https://github.com/ms178/archpkgbuilds/blob/main/packages/alsa-lib-main/PKG… 2. Uncomment the patch that I apply to workaround the issue. 3. Configure the build to use Clang-21 (`bfa7edcc6652bdb37d53e0cec64926aab3f280eb`). 4. Attempt to compile the library. **Expected Behavior:** The compilation should complete successfully without errors. **Actual Behavior:** The compilation fails with multiple errors similar to the following: ``` In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:170:27: error: expected member name or ';' after declaration specifiers 170 | snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:55:27: note: expanded from macro 'note_on' 55 | #define note_on 0x90 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:170:26: error: expected ';' at end of declaration list 170 | snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */ | ^ | ; ../include/alsa/ump_msg.h:171:27: error: expected member name or ';' after declaration specifiers 171 | snd_ump_msg_midi1_note_t note_off; /**< MIDI1 note-off message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:54:27: note: expanded from macro 'note_off' 54 | #define note_off 0x80 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:171:26: error: expected ';' at end of declaration list 171 | snd_ump_msg_midi1_note_t note_off; /**< MIDI1 note-off message */ | ^ | ; ../include/alsa/ump_msg.h:173:26: error: expected member name or ';' after declaration specifiers 173 | snd_ump_msg_midi1_cc CCLD rawmidi _t control_change; /**< MIDI1 control-change message */ | ~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:57:28: note: expanded from macro 'control_change' 57 | #define control_change 0xb0 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h CCLD oldapi :31: ../include/alsa/ump_msg.h:173:24: error: expected ';' at end of declaration list 173 | snd_ump_msg_midi1_cc_t control_change; /**< MIDI1 control-change message */ | ^ | ; ../include/alsa/ump_msg.h:432:27: error: expected member name or ';' after declaration specifiers 432 | snd_ump_msg_midi2_note_t note_on; /**< MIDI2 note-on message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:55:27: note: expanded from macro 'note_on' 55 | #define note_on 0x90 | ^ CCLD pcm_min In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:432:26: error: expected ';' at end of declaration list 432 | snd_ump_msg_midi2_note_t note_on; /**< MIDI2 note-on message */ | ^ | ; ../include/alsa/ump_msg.h:433:27: error: expected member name or ';' after declaration specifiers 433 | snd_ump_msg_midi2_note_t note_off; /**< MIDI2 note-off message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:54:27: note: expanded from macro 'note_off' 54 | #define note_off 0x80 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:433:26: error: expected ';' at end of declaration list 433 | snd_ump_msg_midi2_note_t note_off; /**< MIDI2 note-off message */ | ^ | ; ../include/alsa/ump_msg.h:438:26: error: expected member name or ';' after declaration specifiers 438 | snd_ump_msg_midi2_cc_t control_change; /**< MIDI2 control-change message */ | ~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:57:28: note: expanded from macro 'control_change' 57 | #define control_change 0xb0 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:438:24: error: expected ';' at end of declaration list 438 | snd_ump_msg_midi2_cc_t control_change; /**< MIDI2 control-change message */ | ^ | ; ../include/alsa/ump_msg.h:711:27: error: expected member name or ';' after declaration specifiers 711 | snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ | ~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.c:57:20: note: expanded from macro 'set_tempo' 57 | #define set_tempo 0x51 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:711:25: error: expected ';' at end of declaration list 711 | snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ | ^ | ; CCLD control CCLD chmap 14 errors generated. ``` **Root Cause analysis by AI:** The macros `note_on`, `note_off`, `control_change`, and `set_tempo` defined in `midifile.h` are conflicting with member names used in structures within `include/alsa/ump_msg.h`. This leads to incorrect code expansion and subsequent compilation errors. **Workaround:** The following patch applied to `include/asoundlib-head.h` allows the compilation to succeed: ```diff --- a/include/asoundlib-head.h +++ b/include/asoundlib-head.h @@ -28,6 +28,11 @@ #ifndef __ASOUNDLIB_H #define __ASOUNDLIB_H +#undef note_on +#undef note_off +#undef control_change +#undef set_tempo + #include <unistd.h> #include <stdio.h> #include <stdlib.h> ``` **Suggested Solutions by AI:** While the provided workaround resolves the immediate compilation issue, a more robust solution might involve: 1. **Renaming conflicting members:** Consider renaming the members in `ump_msg.h` (e.g., `midi1_note_on`, `midi2_note_off`, etc.) to avoid clashes with commonly used MIDI event names. 2. **Namespace/prefix:** If renaming is not feasible, consider using a namespace or prefix for the `ump_msg.h` members to prevent conflicts. 3. **Conditional undef:** Instead of unconditionally undefining the macros in `asoundlib-head.h`, a more targeted approach could be to undefine them only within the scope of `ump_msg.h`'s inclusion. 4. **Review macro definitions:** Investigate if the macros in `midifile.h` are strictly necessary or if they can be replaced with constants or enums within a namespace. **Additional Notes:** * I am not a programmer and cannot provide a definitive fix. The provided workaround is based on AI feedback. Issue URL : https://github.com/alsa-project/alsa-lib/issues/436 Repository URL: https://github.com/alsa-project/alsa-lib

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