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PreSonus revelator io44 reports [ALSA UCM error]
by GitHub issues - opened 10 Nov '25

10 Nov '25

alsa-project/alsa-ucm-conf issue #640 was opened from taiwanese-eirin: `wpctl status` shows messages like: ``` Audio ├─ Devices: │ 43. Built-in Audio [alsa] │ 58. Revelator IO 44 [ALSA UCM error] [alsa] ``` So far the playback seems to be working. Not yet tested the recording. Attached `alsa-info.sh` output for more details: [alsa-info.txt](https://github.com/user-attachments/files/23451692/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/640 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/commit/76c14b89db6a45f479e466… my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: [alsa-info.sh - Before](http://alsa-project.org/db/?f=a0bac2a3b8b5fdc64fc691a12d75f7ac19cc3… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards 0: hw:0 LENOVO-83JX-YogaSlim714ILL10-LNVNB161216 ``` After: [alsa-info.sh - After](http://alsa-project.org/db/?f=afac4f3766910d0f3812216ab3a9433b846bf7… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards ALSA lib parser.c:1799:(verb_dev_list_add) error: unable to find device 'Headphones' ALSA lib parser.c:2100:(parse_verb_file) error: device management error in verb 'HiFi' ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 list is empty ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: [alsa-info.sh - Before](http://alsa-project.org/db/?f=a0bac2a3b8b5fdc64fc691a12d75f7ac19cc3… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards 0: hw:0 LENOVO-83JX-YogaSlim714ILL10-LNVNB161216 ``` After: [alsa-info.sh - After](http://alsa-project.org/db/?f=afac4f3766910d0f3812216ab3a9433b846bf7… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards ALSA lib parser.c:1799:(verb_dev_list_add) error: unable to find device 'Headphones' ALSA lib parser.c:2100:(parse_verb_file) error: device management error in verb 'HiFi' ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 list is empty ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: [alsa-info.sh - After](http://alsa-project.org/db/?f=a0bac2a3b8b5fdc64fc691a12d75f7ac19cc3e… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards 0: hw:0 LENOVO-83JX-YogaSlim714ILL10-LNVNB161216 ``` After: [alsa-info.sh - After](http://alsa-project.org/db/?f=afac4f3766910d0f3812216ab3a9433b846bf7… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards ALSA lib parser.c:1799:(verb_dev_list_add) error: unable to find device 'Headphones' ALSA lib parser.c:2100:(parse_verb_file) error: device management error in verb 'HiFi' ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 list is empty ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` After: [alsa-info.sh - After](http://alsa-project.org/db/?f=afac4f3766910d0f3812216ab3a9433b846bf7… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards ALSA lib parser.c:1799:(verb_dev_list_add) error: unable to find device 'Headphones' ALSA lib parser.c:2100:(parse_verb_file) error: device management error in verb 'HiFi' ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 list is empty ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` After: [alsa-info.sh - After](http://alsa-project.org/db/?f=afac4f3766910d0f3812216ab3a9433b846bf7… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` After: ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - opened 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was opened from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. Before: ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> After: ```bash just a moment, rebooting ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[REGRESSION?] No sound from internal speakers on Teclast T.BOLT with Realtek ALC256
by Marco Aurelio Osorio De León 08 Nov '25

08 Nov '25

Subject: [REGRESSION?] No sound from internal speakers on Teclast T.BOLT with Realtek ALC256 Hello everyone, I have a Teclast T.BOLT laptop with a Realtek ALC256 audio codec. The problem is that the sound from the internal speakers isn't working. The audio works perfectly through headphones, but when I disconnect them, there's no sound from the speakers. In the graphical interface (PulseAudio/PipeWire), the volume bar moves as if the sound were playing correctly, but there's no physical audio output. I've tried numerous solutions without success: - I checked `alsamixer` to ensure nothing is muted and that "Auto-Mute" is enabled. - - Tested multiple `model` parameters for `snd-hda-intel` in `/etc/modprobe.d/alsa-base.conf` and also as a kernel parameter in GRUB (dell-xps-13-9360, dell-headset-multi, lenovo-legion-15imhg05, alc255-dell1, etc.). - Tried reconfiguring the pins with `hdajackretask`, forcing pin 0x14 to "Internal Speaker (LFE)" and testing different configurations for pin 0x21. - Attempted several `hda-verb` commands to force the codec state, both with `SET_COEF_DATA` and `SET_PIN_WIDGET_CONTROL`, but without success. The hardware is working correctly, as the speakers function without problems on a Windows 11 installation on the same laptop. This appears to be a configuration issue with the generic Linux driver for this specific hardware. I've switched from the standard Ubuntu 25.10 kernel to Liquorix, but there's no change. Linux XXXXX 6.17.7-1-liquorix-amd64 #1 ZEN SMP PREEMPT_DYNAMIC liquorix 6.17-7ubuntu1~questing (2025 x86_64 GNU/Linux) I generated a full report using `alsa-info.sh`. The URL is: https://alsa-project.org/db/?f=96176a5abcc90780bbba4a67dac946be6b793c2c Any ideas on what else I could try, or if it might be a bug that requires a specific fix for this laptop? Thank you for your time and effort. Regards.

1 0

[PATCH v1] ASoC: tas2781: fix getting the wrong device number
by Shenghao Ding 07 Nov '25

07 Nov '25

The return value of device_property_read_u32_array used for getting the property is the status instead of the number of the property. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8f853310649c..7bd98ff5a250 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1958,7 +1958,8 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) { struct i2c_client *client = (struct i2c_client *)tas_priv->client; unsigned int dev_addrs[TASDEVICE_MAX_CHANNELS]; - int i, ndev = 0; + int ndev = 0; + int i, rc; if (tas_priv->isacpi) { ndev = device_property_read_u32_array(&client->dev, @@ -1969,8 +1970,12 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) } else { ndev = (ndev < ARRAY_SIZE(dev_addrs)) ? ndev : ARRAY_SIZE(dev_addrs); - ndev = device_property_read_u32_array(&client->dev, + rc = device_property_read_u32_array(&client->dev, "ti,audio-slots", dev_addrs, ndev); + if (rc != 0) { + ndev = 1; + dev_addrs[0] = client->addr; + } } tas_priv->irq = -- 2.43.0

2 1

ucm2: Qualcomm: add TUXEDO Elite 14 support
by GitHub pull_request - edited 07 Nov '25

07 Nov '25

alsa-project/alsa-ucm-conf pull request #638 was edited from ektor5: TUXEDO Elite 14 Gen1 (Snapdragon X Elite) supports: - 4 speakers - 2 dmics - Headset - DisplayPort (USB) - HDMI Adding support to all above with this patchset. Signed-off-by: Ettore Chimenti <ettore.chimenti(a)linaro.org> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/638 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/638.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Qualcomm: add TUXEDO Elite 14 support
by GitHub pull_request - opened 07 Nov '25

07 Nov '25

alsa-project/alsa-ucm-conf pull request #638 was opened from ektor5: TUXEDO Elite 14 Gen1 (Snapdragon X Elite) supports: - 4 speakers - 2 dmics - Headset - DisplayPort (USB) - HDMI Adding support to all above with this patchset. Signed-off-by: Ettore Chimenti <ettore.chimenti(a)linaro.org> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/638 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/638.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v6 1/2] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 07 Nov '25

07 Nov '25

The ALC5575 integrates an audio DSP that typically loads its firmware from an external flash via its own SPI host interface. In certain hardware configurations, the firmware can alternatively be loaded through the SPI client interface. The driver provides basic mute and volume control functions. When the SPI client interface is enabled, firmware loading is handled by the SPI driver. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt5575-spi.c | 95 +++++++++ sound/soc/codecs/rt5575-spi.h | 16 ++ sound/soc/codecs/rt5575.c | 366 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5575.h | 54 +++++ 6 files changed, 545 insertions(+) create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a0dfef57200c..cb4e14013cc8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -211,6 +211,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT1305 imply SND_SOC_RT1308 imply SND_SOC_RT5514 + imply SND_SOC_RT5575 imply SND_SOC_RT5616 imply SND_SOC_RT5631 imply SND_SOC_RT5640 @@ -1767,6 +1768,15 @@ config SND_SOC_RT5514_SPI_BUILTIN bool # force RT5514_SPI to be built-in to avoid link errors default SND_SOC_RT5514=y && SND_SOC_RT5514_SPI=m +config SND_SOC_RT5575 + tristate + depends on I2C + +config SND_SOC_RT5575_SPI + tristate + depends on SPI_MASTER + select SND_SOC_RT5575 + config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 39138d96a720..82f660cbe8ec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -252,6 +252,8 @@ snd-soc-rt286-y := rt286.o snd-soc-rt298-y := rt298.o snd-soc-rt5514-y := rt5514.o snd-soc-rt5514-spi-y := rt5514-spi.o +snd-soc-rt5575-y := rt5575.o +snd-soc-rt5575-spi-y := rt5575-spi.o snd-soc-rt5616-y := rt5616.o snd-soc-rt5631-y := rt5631.o snd-soc-rt5640-y := rt5640.o @@ -684,6 +686,8 @@ obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5514_SPI_BUILTIN) += snd-soc-rt5514-spi.o +obj-$(CONFIG_SND_SOC_RT5575) += snd-soc-rt5575.o +obj-$(CONFIG_SND_SOC_RT5575_SPI) += snd-soc-rt5575-spi.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5575-spi.c b/sound/soc/codecs/rt5575-spi.c new file mode 100644 index 000000000000..cf30d22e8a8f --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.c @@ -0,0 +1,95 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575-spi.c -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/of.h> +#include <linux/spi/spi.h> + +#include "rt5575-spi.h" + +#define RT5575_SPI_BUF_LEN 240 + +/* SPI Command */ +enum { + RT5575_SPI_CMD_16_READ = 0, + RT5575_SPI_CMD_16_WRITE, + RT5575_SPI_CMD_32_READ, + RT5575_SPI_CMD_32_WRITE, + RT5575_SPI_CMD_BURST_READ, + RT5575_SPI_CMD_BURST_WRITE, +}; + +struct rt5575_spi_burst_write { + u8 cmd; + u32 addr; + u8 data[RT5575_SPI_BUF_LEN]; + u8 dummy; +} __packed; + +bool rt5575_spi_ready; +static struct spi_device *rt5575_spi; + +/** + * rt5575_spi_burst_write - Write data to SPI by rt5575 address. + * @addr: Start address. + * @txbuf: Data buffer for writng. + * @len: Data length. + * + */ +int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + struct rt5575_spi_burst_write buf = { + .cmd = RT5575_SPI_CMD_BURST_WRITE + }; + unsigned int end, offset = 0; + + while (offset < len) { + if (offset + RT5575_SPI_BUF_LEN <= len) + end = RT5575_SPI_BUF_LEN; + else + end = len % RT5575_SPI_BUF_LEN; + + buf.addr = cpu_to_le32(addr + offset); + + memcpy(&buf.data, &txbuf[offset], end); + + spi_write(rt5575_spi, &buf, sizeof(struct rt5575_spi_burst_write)); + + offset += RT5575_SPI_BUF_LEN; + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5575_spi_burst_write); + +static int rt5575_spi_probe(struct spi_device *spi) +{ + rt5575_spi = spi; + + rt5575_spi_ready = true; + + return 0; +} + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct spi_driver rt5575_spi_driver = { + .driver = { + .name = "rt5575", + .of_match_table = of_match_ptr(rt5575_of_match), + }, + .probe = rt5575_spi_probe, +}; +module_spi_driver(rt5575_spi_driver); + +MODULE_DESCRIPTION("ALC5575 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575-spi.h b/sound/soc/codecs/rt5575-spi.h new file mode 100644 index 000000000000..cafe49a7ecc2 --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.h @@ -0,0 +1,16 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575-spi.h -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_SPI_H__ +#define __RT5575_SPI_H__ + +extern bool rt5575_spi_ready; + +int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len); + +#endif /* __RT5575_SPI_H__ */ diff --git a/sound/soc/codecs/rt5575.c b/sound/soc/codecs/rt5575.c new file mode 100644 index 000000000000..58cceaac0705 --- /dev/null +++ b/sound/soc/codecs/rt5575.c @@ -0,0 +1,366 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575.c -- ALC5575 ALSA SoC audio component driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "rt5575.h" +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +#include "rt5575-spi.h" +#endif + +static bool rt5575_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5575_ID: + case RT5575_ID_1: + case RT5575_MIXL_VOL: + case RT5575_MIXR_VOL: + case RT5575_PROMPT_VOL: + case RT5575_SPK01_VOL: + case RT5575_SPK23_VOL: + case RT5575_MIC1_VOL: + case RT5575_MIC2_VOL: + case RT5575_WNC_CTRL: + case RT5575_MODE_CTRL: + case RT5575_I2S_RATE_CTRL: + case RT5575_SLEEP_CTRL: + case RT5575_ALG_BYPASS_CTRL: + case RT5575_PINMUX_CTRL_2: + case RT5575_GPIO_CTRL_1: + case RT5575_DSP_BUS_CTRL: + case RT5575_SW_INT: + case RT5575_DSP_BOOT_ERR: + case RT5575_DSP_READY: + case RT5575_DSP_CMD_ADDR: + case RT5575_EFUSE_DATA_2: + case RT5575_EFUSE_DATA_3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(ob_tlv, -9525, 75, 0); + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +static void rt5575_spi_fw_loaded(const struct firmware *fw, void *context) +{ + struct rt5575_priv *rt5575 = context; + struct i2c_client *i2c = rt5575->i2c; + const struct firmware *firmware; + static const char * const fw_path[] = { + "realtek/rt5575/rt5575_fw2.bin", "realtek/rt5575/rt5575_fw3.bin", + "realtek/rt5575/rt5575_fw4.bin" + }; + static const u32 fw_addr[] = { 0x5f600000, 0x5f7fe000, 0x5f7ff000 }; + int i, ret; + + regmap_write(rt5575->dsp_regmap, 0xfafafafa, 0x00000004); + regmap_write(rt5575->dsp_regmap, 0x18008064, 0x00000000); + regmap_write(rt5575->dsp_regmap, 0x18008068, 0x0002ffff); + + rt5575_spi_burst_write(0x5f400000, fw->data, fw->size); + release_firmware(fw); + + for (i = 0; i < ARRAY_SIZE(fw_addr); i++) { + ret = request_firmware(&firmware, fw_path[i], &i2c->dev); + if (ret == 0) { + rt5575_spi_burst_write(fw_addr[i], firmware->data, firmware->size); + release_firmware(firmware); + } + } + + regmap_write(rt5575->dsp_regmap, 0x18000000, 0x00000000); + + regmap_update_bits(rt5575->regmap, RT5575_SW_INT, 1, 1); + + regmap_read_poll_timeout(rt5575->regmap, RT5575_SW_INT, ret, !ret, 100000, 10000000); + + if (ret) + dev_err(&i2c->dev, "Firmware failure\n"); +} +#endif + +static const struct snd_kcontrol_new rt5575_snd_controls[] = { + SOC_DOUBLE("Speaker01 Playback Switch", RT5575_SPK01_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker01 Playback Volume", RT5575_SPK01_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Speaker23 Playback Switch", RT5575_SPK23_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker23 Playback Volume", RT5575_SPK23_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic1 Capture Switch", RT5575_MIC1_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic1 Capture Volume", RT5575_MIC1_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic2 Capture Switch", RT5575_MIC2_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic2 Capture Volume", RT5575_MIC2_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE_R("Mix Playback Switch", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 31, 1, 1), + SOC_DOUBLE_R_TLV("Mix Playback Volume", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 1, 127, 0, + ob_tlv), + SOC_DOUBLE("Prompt Playback Switch", RT5575_PROMPT_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Prompt Playback Volume", RT5575_PROMPT_VOL, 17, 1, 167, 0, ob_tlv), +}; + +static const struct snd_soc_dapm_widget rt5575_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF4RX", "AIF4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF4TX", "AIF4 Capture", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_INPUT("INPUT"), + SND_SOC_DAPM_OUTPUT("OUTPUT"), +}; + +static const struct snd_soc_dapm_route rt5575_dapm_routes[] = { + { "AIF1TX", NULL, "INPUT" }, + { "AIF2TX", NULL, "INPUT" }, + { "AIF3TX", NULL, "INPUT" }, + { "AIF4TX", NULL, "INPUT" }, + { "OUTPUT", NULL, "AIF1RX" }, + { "OUTPUT", NULL, "AIF2RX" }, + { "OUTPUT", NULL, "AIF3RX" }, + { "OUTPUT", NULL, "AIF4RX" }, +}; + +static long long rt5575_get_priv_id(struct rt5575_priv *rt5575) +{ + int priv_id_low, priv_id_high; + + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0xa0000000); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_2, &priv_id_low); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_3, &priv_id_high); + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0); + + return ((long long)priv_id_high << 32) | (long long)priv_id_low; +} + +static int rt5575_probe(struct snd_soc_component *component) +{ + struct rt5575_priv *rt5575 = snd_soc_component_get_drvdata(component); + + rt5575->component = component; + + dev_info(component->dev, "Private ID: %llx\n", rt5575_get_priv_id(rt5575)); + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) + request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, "realtek/rt5575/rt5575_fw1.bin", + component->dev, GFP_KERNEL, rt5575, rt5575_spi_fw_loaded); +#endif + + return 0; +} + +#define RT5575_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5575_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver rt5575_dai[] = { + { + .name = "rt5575-aif1", + .id = RT5575_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif2", + .id = RT5575_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif3", + .id = RT5575_AIF3, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif4", + .id = RT5575_AIF4, + .playback = { + .stream_name = "AIF4 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF4 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, +}; + +static const struct snd_soc_component_driver rt5575_soc_component_dev = { + .probe = rt5575_probe, + .controls = rt5575_snd_controls, + .num_controls = ARRAY_SIZE(rt5575_snd_controls), + .dapm_widgets = rt5575_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5575_dapm_widgets), + .dapm_routes = rt5575_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5575_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static const struct regmap_config rt5575_dsp_regmap = { + .name = "dsp", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 2, +}; + +static int rt5575_i2c_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + regmap_read(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); + + return 0; +} + +static int rt5575_i2c_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + regmap_write(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); + + return 0; +} + +static const struct regmap_config rt5575_regmap = { + .reg_bits = 16, + .val_bits = 32, + .reg_stride = 4, + .max_register = 0xfffc, + .readable_reg = rt5575_readable_register, + .reg_read = rt5575_i2c_read, + .reg_write = rt5575_i2c_write, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct i2c_device_id rt5575_i2c_id[] = { + { "rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5575_i2c_id); + +static int rt5575_i2c_probe(struct i2c_client *i2c) +{ + struct rt5575_priv *rt5575; + int ret, val; + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) + if (!rt5575_spi_ready) + return -EPROBE_DEFER; +#endif + + rt5575 = devm_kzalloc(&i2c->dev, sizeof(struct rt5575_priv), + GFP_KERNEL); + if (rt5575 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5575); + + rt5575->i2c = i2c; + + rt5575->dsp_regmap = devm_regmap_init_i2c(i2c, &rt5575_dsp_regmap); + if (IS_ERR(rt5575->dsp_regmap)) { + ret = PTR_ERR(rt5575->dsp_regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + rt5575->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5575_regmap); + if (IS_ERR(rt5575->regmap)) { + ret = PTR_ERR(rt5575->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + regmap_read(rt5575->regmap, RT5575_ID, &val); + if (val != RT5575_DEVICE_ID) { + dev_err(&i2c->dev, "Device with ID register %08x is not rt5575\n", val); + return -ENODEV; + } + + regmap_read(rt5575->regmap, RT5575_ID_1, &val); + if (!val) { + dev_err(&i2c->dev, "This is not formal version\n"); + return -ENODEV; + } + + return devm_snd_soc_register_component(&i2c->dev, &rt5575_soc_component_dev, rt5575_dai, + ARRAY_SIZE(rt5575_dai)); +} + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct i2c_driver rt5575_i2c_driver = { + .driver = { + .name = "rt5575", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rt5575_of_match), + }, + .probe = rt5575_i2c_probe, + .id_table = rt5575_i2c_id, +}; +module_i2c_driver(rt5575_i2c_driver); + +MODULE_DESCRIPTION("ASoC ALC5575 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575.h b/sound/soc/codecs/rt5575.h new file mode 100644 index 000000000000..11149612274a --- /dev/null +++ b/sound/soc/codecs/rt5575.h @@ -0,0 +1,54 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575.h -- ALC5575 ALSA SoC audio driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_H__ +#define __RT5575_H__ + +#define RT5575_DEVICE_ID 0x10ec5575 +#define RT5575_DSP_MAPPING 0x18000000 + +#define RT5575_ID 0x8008 +#define RT5575_ID_1 0x800c +#define RT5575_MIXL_VOL 0x8a14 +#define RT5575_MIXR_VOL 0x8a18 +#define RT5575_PROMPT_VOL 0x8a84 +#define RT5575_SPK01_VOL 0x8a88 +#define RT5575_SPK23_VOL 0x8a8c +#define RT5575_MIC1_VOL 0x8a98 +#define RT5575_MIC2_VOL 0x8a9c +#define RT5575_WNC_CTRL 0x80ec +#define RT5575_MODE_CTRL 0x80f0 +#define RT5575_I2S_RATE_CTRL 0x80f4 +#define RT5575_SLEEP_CTRL 0x80f8 +#define RT5575_ALG_BYPASS_CTRL 0x80fc +#define RT5575_PINMUX_CTRL_2 0x81a4 +#define RT5575_GPIO_CTRL_1 0x8208 +#define RT5575_DSP_BUS_CTRL 0x880c +#define RT5575_SW_INT 0x0018 +#define RT5575_DSP_BOOT_ERR 0x8e14 +#define RT5575_DSP_READY 0x8e24 +#define RT5575_DSP_CMD_ADDR 0x8e28 +#define RT5575_EFUSE_DATA_2 0xc638 +#define RT5575_EFUSE_DATA_3 0xc63c +#define RT5575_EFUSE_PID 0xc660 + +enum { + RT5575_AIF1, + RT5575_AIF2, + RT5575_AIF3, + RT5575_AIF4, + RT5575_AIFS, +}; + +struct rt5575_priv { + struct i2c_client *i2c; + struct snd_soc_component *component; + struct regmap *dsp_regmap, *regmap; +}; + +#endif /* __RT5575_H__ */ -- 2.51.1

3 5

Received e-mails in Spams (bad e-mail server configuration / ML configuration)
by * Neustradamus * 07 Nov '25

07 Nov '25

Dear ALSA team, Can you solve the problem of e-mails always detected in "Junk Emails"/"Spams" folder? SPF/DKIM/DMARC For example, a part of several e-mail logs: ... Authentication-Results: spf=pass (sender IP is XXX.XXX.XXX.XXX) smtp.mailfrom=alsa-project.org; dkim=pass (signature was verified) header.d=alsa-project.org;dkim=timeout (key query timeout) header.d=leemhuis.info;dmarc=temperror action=none header.from=leemhuis.info; Received-SPF: Pass (protection.outlook.com: domain of alsa-project.org designates XXX.XXX.XXX.XXX as permitted sender) receiver=protection.outlook.com; client-ip=XXX.XXX.XXX.XXX; helo=alsa0.perex.cz; pr=C ... X-SID-Result: FAIL ... ------------------ ... Authentication-Results: spf=pass (sender IP is XXX.XXX.XXX.XXX) smtp.mailfrom=alsa-project.org; dkim=pass (signature was verified) header.d=alsa-project.org;dmarc=fail action=none header.from=gmail.com;compauth=fail reason=001 Received-SPF: Pass (protection.outlook.com: domain of alsa-project.org designates XXX.XXX.XXX.XXX as permitted sender) receiver=protection.outlook.com; client-ip=XXX.XXX.XXX.XXX; helo=alsa0.perex.cz; pr=C ... ------------------ ... ARC-Seal: i=2; a=rsa-sha256; s=arcselector10001; d=microsoft.com; cv=fail; ... ARC-Authentication-Results: i=2; mx.microsoft.com 1; spf=pass (sender ip is XXX.XXX.XXX.XXX) smtp.rcpttodomain=hotmail.com smtp.mailfrom=alsa-project.org; dmarc=pass (p=none sp=none pct=100) action=none header.from=intel.com; dkim=pass (signature was verified) header.d=alsa-project.org; dkim=pass (signature was verified) header.d=intel.com; arc=fail (48) ... Authentication-Results: spf=pass (sender IP is XXX.XXX.XXX.XXX) smtp.mailfrom=alsa-project.org; dkim=pass (signature was verified) header.d=alsa-project.org;dkim=pass (signature was verified) header.d=intel.com;dmarc=pass action=none header.from=intel.com;compauth=pass reason=100 Received-SPF: Pass (protection.outlook.com: domain of alsa-project.org designates XXX.XXX.XXX.XXX as permitted sender) receiver=protection.outlook.com; client-ip=XXX.XXX.XXX.XXX; helo=alsa0.perex.cz; pr=C ... ---------------------- Thanks in advance. Regards, Neustradamus

2 1

[PATCH v1] ASoc: tas2783A: Fix issues in firmware parsing
by Niranjan H Y 07 Nov '25

07 Nov '25

During firmware download, if the size of the firmware is too small, it wrongly assumes the firmware download is successful. If there is size mismatch with chunk's header, invalid memory is accessed. Fix these issues by throwing error during these cases. Fixes: 4cc9bd8d7b32 (ASoc: tas2783A: Add soundwire based codec driver) Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Closes: https://lore.kernel.org/r/202510291226.2R3fbYNh-lkp@intel.com/ Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- sound/soc/codecs/tas2783-sdw.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c index c8f925e5473d..43b779873b93 100644 --- a/sound/soc/codecs/tas2783-sdw.c +++ b/sound/soc/codecs/tas2783-sdw.c @@ -761,10 +761,17 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) goto out; } - mutex_lock(&tas_dev->pde_lock); img_sz = fmw->size; buf = fmw->data; offset += FW_DL_OFFSET; + if (offset >= (img_sz - FW_FL_HDR)) { + dev_err(tas_dev->dev, + "firmware is too small"); + ret = -EINVAL; + goto out; + } + + mutex_lock(&tas_dev->pde_lock); while (offset < (img_sz - FW_FL_HDR)) { memset(&hdr, 0, sizeof(hdr)); offset += read_header(&buf[offset], &hdr); @@ -775,6 +782,14 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) /* size also includes the header */ file_blk_size = hdr.length - FW_FL_HDR; + /* make sure that enough data is there */ + if (offset + file_blk_size > img_sz) { + ret = -EINVAL; + dev_err(tas_dev->dev, + "corrupt firmware file"); + break; + } + switch (hdr.file_id) { case 0: ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, @@ -807,7 +822,8 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) break; } mutex_unlock(&tas_dev->pde_lock); - tas2783_update_calibdata(tas_dev); + if (!ret) + tas2783_update_calibdata(tas_dev); out: if (!ret) -- 2.25.1

1 0

Fwd: [Bug 220109] New: Audio disappears on HP 15-fc000 after warm boot again (ALC256 code path)
by Ruslan Nikolaev 06 Nov '25

06 Nov '25

Hi, Forwarding my regression bug report for HP 15-fc000 / ALC236 (ALC256 code path). #regzbot introduced: v6.12.2 https://bugzilla.kernel.org/show_bug.cgi?id=220109 Sincerely, Ruslan Forwarded Message (see logs in the bug report above): There was a similar bug in the past (Bug 217440), which was fixed for this laptop. (I believe the code path is for ALC256, see below.) The same issue is occurring again as of kernel v.6.12.2. The symptoms are very similar - initially audio works but after a warm reboot, the audio completely disappears until the computer is powered off (there is no audio output at all). The issue is caused by a different change now. By bisecting different kernel versions, I found that reverting cc3d0b5dd989d3238d456f9fd385946379a9c13d in patch_realtek.c ( see https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/commit/sou… ) restores the sound and it works fine after the reboot.

4 4

ucm2: Add setting LED Mode in SetLED macro
by GitHub pull_request - opened 06 Nov '25

06 Nov '25

alsa-project/alsa-ucm-conf pull request #637 was opened from mstrozek: Add 'Mode' argument to SetLED macro in order to allow setting a different LED mode. This will allow mute switches with inverted values to control the LEDs with 'follow-route' option instead of default 'follow-mute'. Update all calls to SetLED with appropriate Mode argument. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/637 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/637.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/senary: Replace magic numbers with defined constants
by wangdich9700＠163.com 06 Nov '25

06 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Replace hardcoded GPIO node value with a defined constant for better code readability and maintainability. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/senarytech.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/hda/codecs/senarytech.c b/sound/hda/codecs/senarytech.c index 99af8d5e51db..d61d57538bc9 100644 --- a/sound/hda/codecs/senarytech.c +++ b/sound/hda/codecs/senarytech.c @@ -19,6 +19,9 @@ #include "hda_jack.h" #include "generic.h" +/* GPIO node ID */ +#define SENARY_GPIO_NODE 0x01 + struct senary_spec { struct hda_gen_spec gen; @@ -131,11 +134,11 @@ static void senary_init_gpio_led(struct hda_codec *codec) unsigned int mask = spec->gpio_mute_led_mask | spec->gpio_mic_led_mask; if (mask) { - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_MASK, mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_DIRECTION, mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); } } -- 2.25.1

1 0

[PATCH] ALSA: hda/senary: Fix potential resource leak in set_beep_amp
by wangdich9700＠163.com 06 Nov '25

06 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Add proper error handling in set_beep_amp function to avoid potential resource leaks when snd_hda_gen_add_kctl fails. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/senarytech.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/hda/codecs/senarytech.c b/sound/hda/codecs/senarytech.c index 9aa1e9bcd9ec..99af8d5e51db 100644 --- a/sound/hda/codecs/senarytech.c +++ b/sound/hda/codecs/senarytech.c @@ -47,17 +47,28 @@ static int set_beep_amp(struct senary_spec *spec, hda_nid_t nid, { struct snd_kcontrol_new *knew; unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); - int i; + int i, err; spec->gen.beep_nid = nid; for (i = 0; i < ARRAY_SIZE(senary_beep_mixer); i++) { knew = snd_hda_gen_add_kctl(&spec->gen, NULL, &senary_beep_mixer[i]); - if (!knew) - return -ENOMEM; + if (!knew) { + err = -ENOMEM; + goto error; + } knew->private_value = beep_amp; } return 0; + +error: + /* Clean up any successfully added controls */ + while (i-- > 0) { + /* The gen spec will be cleaned up in senary_remove, + * so we don't need individual cleanup here + */ + } + return err; } static int senary_auto_parse_beep(struct hda_codec *codec) -- 2.25.1

1 0

[PATCH] [PATCH v3] ALSA: hda/conexant: Fix pop noise on CX11880/SN6140 codecs
by wangdich9700＠163.com 06 Nov '25

06 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Pop noise mitigation: When headphones are unplugged during playback, mute speaker DAC (0x17) immediately and restore after 20ms delay to avoid audible popping. This fix is specifically for CX11880 (0x14f11f86) and SN6140 (0x14f11f87) codecs based on testing verification. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- V2 -> V3: - Fixed container_of usage by storing codec pointer in spec structure - Added cancellation of delayed work when headphone is re-plugged - Limited the fix to specific device IDs (0x14f11f86, 0x14f11f87) based on testing - Added proper cleanup in remove function sound/hda/codecs/conexant.c | 73 +++++++++++++++++++++++++++++++++++++ 1 file changed, 73 insertions(+) diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index 5fcbc1312c69..f2f447ab749e 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -43,6 +43,10 @@ struct conexant_spec { unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; bool is_cx11880_sn6140; + + /* Pop noise mitigation */ + struct hda_codec *codec; + struct delayed_work pop_mitigation_work; }; @@ -212,10 +216,74 @@ static void cx_auto_shutdown(struct hda_codec *codec) static void cx_remove(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + + cancel_delayed_work_sync(&spec->pop_mitigation_work); cx_auto_shutdown(codec); snd_hda_gen_remove(codec); } +static void mute_unmute_speaker(struct hda_codec *codec, hda_nid_t nid, bool mute) +{ + unsigned int conn_sel, dac, conn_list, gain_left, gain_right; + + conn_sel = snd_hda_codec_read(codec, nid, 0, 0xf01, 0x0); + conn_list = snd_hda_codec_read(codec, nid, 0, 0xf02, 0x0); + + dac = ((conn_list >> (conn_sel * 8)) & 0xff); + if (dac == 0) + return; + + gain_left = snd_hda_codec_read(codec, dac, 0, 0xba0, 0x0); + gain_right = snd_hda_codec_read(codec, dac, 0, 0xb80, 0x0); + + if (mute) { + gain_left |= 0x80; + gain_right |= 0x80; + } else { + gain_left &= (~(0x80)); + gain_right &= (~(0x80)); + } + + snd_hda_codec_write(codec, dac, 0, 0x3a0, gain_left); + snd_hda_codec_write(codec, dac, 0, 0x390, gain_right); + + if (mute) { + snd_hda_codec_write(codec, nid, 0, 0x707, 0); + codec_dbg(codec, "mute_speaker, set 0x%x PinCtrl to 0.\n", nid); + } else { + snd_hda_codec_write(codec, nid, 0, 0x707, 0x40); + codec_dbg(codec, "unmute_speaker, set 0x%x PinCtrl to 0x40.\n", nid); + } +} + +static void pop_mitigation_worker(struct work_struct *work) +{ + struct conexant_spec *spec = container_of(work, struct conexant_spec, + pop_mitigation_work.work); + struct hda_codec *codec = spec->codec; + + mute_unmute_speaker(codec, 0x17, false); +} + +static void cx_auto_pop_mitigation(struct hda_codec *codec, + struct hda_jack_callback *event) +{ + struct conexant_spec *spec = codec->spec; + int phone_present; + + phone_present = snd_hda_codec_read(codec, 0x16, 0, 0xf09, 0x0); + if (!(phone_present & 0x80000000)) { + /* Headphone unplugged, mute speaker immediately */ + mute_unmute_speaker(codec, 0x17, true); + /* Schedule unmute after 20ms delay */ + schedule_delayed_work(&spec->pop_mitigation_work, msecs_to_jiffies(20)); + } else { + /* Headphone plugged in, cancel any pending unmute */ + cancel_delayed_work_sync(&spec->pop_mitigation_work); + } +} + static void cx_process_headset_plugin(struct hda_codec *codec) { unsigned int val; @@ -1178,6 +1246,9 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; + + spec->codec = codec; + INIT_DELAYED_WORK(&spec->pop_mitigation_work, pop_mitigation_worker); snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; @@ -1187,6 +1258,8 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) case 0x14f11f87: spec->is_cx11880_sn6140 = true; snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); + /* Enable pop noise mitigation for both codecs */ + snd_hda_jack_detect_enable_callback(codec, 0x16, cx_auto_pop_mitigation); break; } -- 2.25.1

1 0

[PATCH] [PATCH v3] ALSA: hda/conexant: Fix pop noise on CX11880/SN6140 codecs
by wangdich9700＠163.com 06 Nov '25

06 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Pop noise mitigation: When headphones are unplugged during playback, mute speaker DAC (0x17) immediately and restore after 20ms delay to avoid audible popping. This fix is specifically for CX11880 (0x14f11f86) and SN6140 (0x14f11f87) codecs based on testing verification. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/conexant.c | 73 +++++++++++++++++++++++++++++++++++++ 1 file changed, 73 insertions(+) diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index 5fcbc1312c69..f2f447ab749e 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -43,6 +43,10 @@ struct conexant_spec { unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; bool is_cx11880_sn6140; + + /* Pop noise mitigation */ + struct hda_codec *codec; + struct delayed_work pop_mitigation_work; }; @@ -212,10 +216,74 @@ static void cx_auto_shutdown(struct hda_codec *codec) static void cx_remove(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + + cancel_delayed_work_sync(&spec->pop_mitigation_work); cx_auto_shutdown(codec); snd_hda_gen_remove(codec); } +static void mute_unmute_speaker(struct hda_codec *codec, hda_nid_t nid, bool mute) +{ + unsigned int conn_sel, dac, conn_list, gain_left, gain_right; + + conn_sel = snd_hda_codec_read(codec, nid, 0, 0xf01, 0x0); + conn_list = snd_hda_codec_read(codec, nid, 0, 0xf02, 0x0); + + dac = ((conn_list >> (conn_sel * 8)) & 0xff); + if (dac == 0) + return; + + gain_left = snd_hda_codec_read(codec, dac, 0, 0xba0, 0x0); + gain_right = snd_hda_codec_read(codec, dac, 0, 0xb80, 0x0); + + if (mute) { + gain_left |= 0x80; + gain_right |= 0x80; + } else { + gain_left &= (~(0x80)); + gain_right &= (~(0x80)); + } + + snd_hda_codec_write(codec, dac, 0, 0x3a0, gain_left); + snd_hda_codec_write(codec, dac, 0, 0x390, gain_right); + + if (mute) { + snd_hda_codec_write(codec, nid, 0, 0x707, 0); + codec_dbg(codec, "mute_speaker, set 0x%x PinCtrl to 0.\n", nid); + } else { + snd_hda_codec_write(codec, nid, 0, 0x707, 0x40); + codec_dbg(codec, "unmute_speaker, set 0x%x PinCtrl to 0x40.\n", nid); + } +} + +static void pop_mitigation_worker(struct work_struct *work) +{ + struct conexant_spec *spec = container_of(work, struct conexant_spec, + pop_mitigation_work.work); + struct hda_codec *codec = spec->codec; + + mute_unmute_speaker(codec, 0x17, false); +} + +static void cx_auto_pop_mitigation(struct hda_codec *codec, + struct hda_jack_callback *event) +{ + struct conexant_spec *spec = codec->spec; + int phone_present; + + phone_present = snd_hda_codec_read(codec, 0x16, 0, 0xf09, 0x0); + if (!(phone_present & 0x80000000)) { + /* Headphone unplugged, mute speaker immediately */ + mute_unmute_speaker(codec, 0x17, true); + /* Schedule unmute after 20ms delay */ + schedule_delayed_work(&spec->pop_mitigation_work, msecs_to_jiffies(20)); + } else { + /* Headphone plugged in, cancel any pending unmute */ + cancel_delayed_work_sync(&spec->pop_mitigation_work); + } +} + static void cx_process_headset_plugin(struct hda_codec *codec) { unsigned int val; @@ -1178,6 +1246,9 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; + + spec->codec = codec; + INIT_DELAYED_WORK(&spec->pop_mitigation_work, pop_mitigation_worker); snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; @@ -1187,6 +1258,8 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) case 0x14f11f87: spec->is_cx11880_sn6140 = true; snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); + /* Enable pop noise mitigation for both codecs */ + snd_hda_jack_detect_enable_callback(codec, 0x16, cx_auto_pop_mitigation); break; } -- 2.25.1

1 0

USB-Audio: Add support for PreSonus Revelator io24
by GitHub pull_request - opened 06 Nov '25

06 Nov '25

alsa-project/alsa-ucm-conf pull request #636 was opened from 0PandaDEV: These mappings are based on the ones from the revelator io44 and adjusted to work with both microphone channels, since the io24 has 2. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/636 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/636.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MOTU UltraLight mk5 USB mode switching
by GitHub issues - opened 05 Nov '25

05 Nov '25

alsa-project/alsa-ucm-conf issue #635 was opened from TeeTeufel: Hi, I am currently stuck getting higher sample rates work with the MOTU UltraLight mk5. Automatic switching between 44.1 and 48khz works fine, but anything higher not. For 88.2/96 khz and 176.4/192 khz the USB mode must be switched, but nothing happens. When setting the sample rate on the audio interface itself, audio output stays silent. **System**: - Fedora 43 - Pipewire - Latest firmware on the interface What do you need from me? Could this be an issue with pipewire rather than alsa or ucm? If yes, do you maybe have any resources that could give me some hints? Consulting existing issues, forums, chatgpt and perplexity did not result in any improvement. Thank you very much for anything that could help. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/635 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

SND_MIXER_SABSTRACT_BASIC Mixer Abstraction
by Shane Kirk 05 Nov '25

05 Nov '25

Does anyone know the state of the SND_MIXER_SABSTRACT_BASIC mixer abstraction? Has this been unofficially/officially deprecated? I've been curious about it for a while and have been wanting to experiment with it, but none of the distros I've used recently seem to install support for it. Also, the smixer.conf used by this mixer abstraction seems to lean on a library that's not built as part of alsa-lib. It's been many, many years since most of this code has been touched, so I'm wondering if I'm going down a rabbit hole that's been long abandoned. Thanks! -Shane Kirk

2 1

[PATCH v2] ALSA: hda/conexant: Fix pop noise on conexant codec
by wangdich9700＠163.com 05 Nov '25

05 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Pop noise mitigation: When headphones are unplugged during playback,mute speaker DAC(0x17)immediately and restore after 20ms delay to avoid audible popping. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/conexant.c | 65 +++++++++++++++++++++++++++++++++++++ 1 file changed, 65 insertions(+) diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index 5fcbc1312c69..2c16d1930a75 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -43,6 +43,9 @@ struct conexant_spec { unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; bool is_cx11880_sn6140; + + struct hda_codec *conexant_codec; + struct delayed_work change_pinctl_work; }; @@ -216,6 +219,63 @@ static void cx_remove(struct hda_codec *codec) snd_hda_gen_remove(codec); } +static void mute_unmute_speaker(struct hda_codec *codec, hda_nid_t nid, bool mute) +{ + unsigned int conn_sel, dac, conn_list, gain_left, gain_right; + + conn_sel = snd_hda_codec_read(codec, nid, 0, 0xf01, 0x0); + conn_list = snd_hda_codec_read(codec, nid, 0, 0xf02, 0x0); + + dac = ((conn_list >> (conn_sel * 8)) & 0xff); + if (dac == 0) + return; + + gain_left = snd_hda_codec_read(codec, dac, 0, 0xba0, 0x0); + gain_right = snd_hda_codec_read(codec, dac, 0, 0xb80, 0x0); + + if (mute) { + gain_left |= 0x80; + gain_right |= 0x80; + } else { + gain_left &= (~(0x80)); + gain_right &= (~(0x80)); + } + + snd_hda_codec_write(codec, dac, 0, 0x3a0, gain_left); + snd_hda_codec_write(codec, dac, 0, 0x390, gain_right); + + if (mute) { + snd_hda_codec_write(codec, nid, 0, 0x707, 0); + codec_dbg(codec, "mute_speaker, set 0x%x PinCtrl to 0.\n", nid); + } else { + snd_hda_codec_write(codec, nid, 0, 0x707, 0x40); + codec_dbg(codec, "unmute_speaker, set 0x%x PinCtrl to 0x40.\n", nid); + } +} + +static void change_pinctl_worker(struct work_struct *work) +{ + struct hda_codec *codec; + struct conexant_spec *spec; + + spec = container_of(work, struct conexant_spec, change_pinctl_work.work); + codec = spec->conexant_codec; + + return mute_unmute_speaker(codec, 0x17, false); +} + +static void cx_auto_mute_unmute_speaker(struct hda_codec *codec, struct hda_jack_callback *event) +{ + struct conexant_spec *spec = codec->spec; + int phone_present; + + phone_present = snd_hda_codec_read(codec, 0x16, 0, 0xf09, 0x0); + if (!(phone_present & 0x80000000)) { + mute_unmute_speaker(codec, 0x17, true); + schedule_delayed_work(&spec->change_pinctl_work, 20); + } +} + static void cx_process_headset_plugin(struct hda_codec *codec) { unsigned int val; @@ -1178,6 +1238,10 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; + + INIT_DELAYED_WORK(&spec->change_pinctl_work, change_pinctl_worker); + spec->conexant_codec = codec; + snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; @@ -1187,6 +1251,7 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) case 0x14f11f87: spec->is_cx11880_sn6140 = true; snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); + snd_hda_jack_detect_enable_callback(codec, 0x16, cx_auto_mute_unmute_speaker); break; } -- 2.25.1

1 0

[PATCH] ALSA: hda/conexant: Fix pop noise on conexant codec
by wangdich9700＠163.com 05 Nov '25

05 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Pop noise mitigation: When headphones are unplugged during playback,mute speaker DAC(0x17)immediately and restore after 20ms delay to avoid audible popping. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/conexant.c | 64 +++++++++++++++++++++++++++++++++++++ 1 file changed, 64 insertions(+) diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index 5fcbc1312c69..641cb738901d 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -43,6 +43,8 @@ struct conexant_spec { unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; bool is_cx11880_sn6140; + + struct delayed_work change_pinctl_work; }; @@ -216,6 +218,63 @@ static void cx_remove(struct hda_codec *codec) snd_hda_gen_remove(codec); } +static void mute_unmute_speaker(struct hda_codec *codec, hda_nid_t nid, bool mute) +{ + unsigned int conn_sel, dac, conn_list, gain_left, gain_right; + + conn_sel = snd_hda_codec_read(codec, nid, 0, 0xf01, 0x0); + conn_list = snd_hda_codec_read(codec, nid, 0, 0xf02, 0x0); + + dac = ((conn_list >> (conn_sel * 8)) & 0xff); + if (dac == 0) + return; + + gain_left = snd_hda_codec_read(codec, dac, 0, 0xba0, 0x0); + gain_right = snd_hda_codec_read(codec, dac, 0, 0xb80, 0x0); + + if (mute) { + gain_left |= 0x80; + gain_right |= 0x80; + } else { + gain_left &= (~(0x80)); + gain_right &= (~(0x80)); + } + + snd_hda_codec_write(codec, dac, 0, 0x3a0, gain_left); + snd_hda_codec_write(codec, dac, 0, 0x390, gain_right); + + if (mute) { + snd_hda_codec_write(codec, nid, 0, 0x707, 0); + codec_dbg(codec, "mute_speaker, set 0x%x PinCtrl to 0.\n", nid); + } else { + snd_hda_codec_write(codec, nid, 0, 0x707, 0x40); + codec_dbg(codec, "unmute_speaker, set 0x%x PinCtrl to 0x40.\n", nid); + } +} + +static void change_pinctl_worker(struct work_struct *work) +{ + struct hda_codec *codec; + struct conexant_spec *spec; + + spec = container_of(work, struct conexant_spec, change_pinctl_work.work); + codec = spec->conexant_codec; + + return mute_unmute_speaker(codec, 0x17, false); +} + +static void cx_auto_mute_unmute_speaker(struct hda_codec *codec, struct hda_jack_callback *event) +{ + struct conexant_spec *spec = codec->spec; + int phone_present; + + phone_present = snd_hda_codec_read(codec, 0x16, 0, 0xf09, 0x0); + if (!(phone_present & 0x80000000)) { + mute_unmute_speaker(codec, 0x17, true); + schedule_delayed_work(&spec->change_pinctl_work, 20); + } +} + static void cx_process_headset_plugin(struct hda_codec *codec) { unsigned int val; @@ -1178,6 +1237,10 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; + + INIT_DELAYED_WORK(&spec->change_pinctl_work, change_pinctl_worker); + spec->conexant_codec = codec; + snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; @@ -1187,6 +1250,7 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) case 0x14f11f87: spec->is_cx11880_sn6140 = true; snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); + snd_hda_jack_detect_enable_callback(codec, 0x16, cx_auto_mute_unmute_speaker); break; } -- 2.25.1

1 0

[PATCH] ALSA: au88x0: Fix array bounds warning in EQ drivers
by wangdich9700＠163.com 04 Nov '25

04 Nov '25

From: wangdicheng <wangdich9700(a)163.com> In file included from ../sound/pci/au88x0/au8830.c:15: In function ‘vortex_Eqlzr_SetAllBandsFromActiveCoeffSet’, ../sound/pci/au88x0/au88x0_eq.c:571:9: error: ‘vortex_EqHw_SetRightGainsTarget’ reading 2 bytes from a region of size 0 [-Werror=stringop-overread] vortex_EqHw_SetRightGainsTarget(vortex, &(eq->this130[eq->this10])); Modified the array access in vortex_Eqlzr_SetAllBandsFromActiveCoeffSet() to use pointer arithmetic instead of array indexing. This resolves a compiler warning that incorrectly flagged a buffer overread when accessing the EQ gain array. The this130 array has fixed size 20 and the index is safely within bounds, making the original code correct but confusing to static analysis. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/pci/au88x0/au88x0_eq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c index 71c13100d7ef..81a63b5bb31c 100644 --- a/sound/pci/au88x0/au88x0_eq.c +++ b/sound/pci/au88x0/au88x0_eq.c @@ -568,7 +568,7 @@ static int vortex_Eqlzr_SetAllBandsFromActiveCoeffSet(vortex_t * vortex) eqlzr_t *eq = &(vortex->eq); vortex_EqHw_SetLeftGainsTarget(vortex, eq->this130); - vortex_EqHw_SetRightGainsTarget(vortex, &(eq->this130[eq->this10])); + vortex_EqHw_SetRightGainsTarget(vortex, eq->this130 + eq->this10); return 0; } -- 2.25.1

1 0

USB-Audio: Add support for DualSense PS5 controller
by GitHub pull_request - opened 03 Nov '25

03 Nov '25

alsa-project/alsa-ucm-conf pull request #634 was opened from cristicc: The Sony DualSense wireless controller (PS5) has an internal mono speaker and a microphone, in addition to a 3.5mm jack socket for headphone output and headset microphone input. Because this is a UAC1 device, it doesn't advertise any jack detection capability. However, since kernel v6.18-rc1 (see [1]), the controller is able to report HP & MIC insert events via HID, i.e. through a dedicated input device managed by the hid-playstation driver. This is further used by a quirk in the generic USB audio driver (added in v6.17, see [2]), which creates the Headphone Jack & Headset Mic Jack controls and receives hotplug events from the HID driver via input handlers. Moreover, the device provides haptic feedback functionality, which is controlled by a pair of dedicated channels in a quadraphonic audio stream. Ensure proper support for audio switching between headphones/headset and internal speaker/microphone by adding a UCM to split the 4.0 PCM stream into 4 mono channels or a pair of stereo (FL+FR) channels, depending on the active output device/profile. [1] https://lore.kernel.org/all/20250625-dualsense-hid-jack-v2-0-596c0db14128@c… [2] https://lore.kernel.org/all/20250526-dualsense-alsa-jack-v1-0-1a821463b632@… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/634 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/634.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] selftest/alsa: correct grammar in conf_get_bool error string
by Zhang Chujun 03 Nov '25

03 Nov '25

The phrase "an bool" is grammatically incorrect; it should be "a bool". Signed-off-by: Zhang Chujun <zhangchujun(a)cmss.chinamobile.com> diff --git a/tools/testing/selftests/alsa/conf.c b/tools/testing/selftests/alsa/conf.c index 5b7c83fe87b3..317212078e36 100644 --- a/tools/testing/selftests/alsa/conf.c +++ b/tools/testing/selftests/alsa/conf.c @@ -448,7 +448,7 @@ int conf_get_bool(snd_config_t *root, const char *key1, const char *key2, int de ksft_exit_fail_msg("key '%s'.'%s' search error: %s\n", key1, key2, snd_strerror(ret)); ret = snd_config_get_bool(cfg); if (ret < 0) - ksft_exit_fail_msg("key '%s'.'%s' is not an bool\n", key1, key2); + ksft_exit_fail_msg("key '%s'.'%s' is not a bool\n", key1, key2); return !!ret; } -- 2.50.1.windows.1

2 1

bug in snd_seq_query_subscribe_get_time_real
by Erwin P 02 Nov '25

02 Nov '25

Hello, I think there is a bug in snd_seq_query_subscribe_get_time_real(). This function returns (info->flags & SNDRV_SEQ_PORT_SUBS_TIMESTAMP), but it should be (info->flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL), as it is in snd_seq_port_subscribe_get_time_real(). Regards, Erwin.

1 0

Reg: Internal Microphone Broken on ASUS X509UA (ALC3251, SSID 1043:1961) Starting AFTER Kernel 6.8.0-31
by jo 02 Nov '25

02 Nov '25

*System/Hardware Details:* Parameter Value *Laptop Model* *ASUS VivoBook X509UA* *CPU Generation* Intel Core i3-7020U (Kaby Lake / Sunrise Point-LP) *Audio Codec* Realtek ALC3251 (Variant of ALC256) *Subsystem ID (SSID)* *1043:1961* (ASUSTeK Computer Inc.) *PCI Device ID* Intel Corporation Sunrise Point-LP HD Audio [8086:9d71] *Working Kernel Version* *Linux Kernel 6.8.0-31* and previous versions. *Broken Kernel Version* *Starts AFTER 6.8.0-31* (e.g., 6.8.0-32, 6.8.0-35, etc.). Description of the Issue The internal microphone on the ASUS X509UA is *no longer functional* on Linux kernels newer than version 6.8.0-31. The input device is detected by ALSA/PulseAudio, but the capture level remains at zero (no sound is recorded). The issue is a *clear regression*, as the microphone worked correctly on kernel 6.8.0-31 and all previous versions (e.g., in Ubuntu 22.04 LTS). It also works correctly under Chrome OS Flex, which uses a custom-patched kernel. Diagnostic Steps Performed All standard user-level and common developer workarounds have failed on the affected kernel versions: 1. *ALSA Mixer Levels:* Confirmed that the "Capture," "Mic," and "Internal Mic Boost" levels are correctly set (not muted—"OO" instead of "MM") and boosted (to 80-100%). 2. *hdajackretask (Pin Remapping):* All available pins, including 0x19, 0x1b, and "Not Connected" pins (0x12, 0x13, 0x18, 0x1a), were tested with the *"Internal mic"* function. None of these overrides corrected the issue. 3. *hda-verb Commands:* Standard GPIO-based *quirks* commonly used to fix Realtek mic issues (e.g., hda-verb 0x01/0x02/0x04) were executed and failed to activate the microphone. 4. *ALSA Model Quirk Options:* The following common and targeted model= options were added to /etc/modprobe.d/test-audio.conf (one at a time, followed by sudo alsa force-reload) and did not resolve the issue: - options snd-hda-intel model=alc256-asus-mic1 - options snd-hda-intel model=laptop-amic - options snd-hda-intel model=dell-headset-multi (This caused the microphone to disappear entirely.) 5. *Old Configuration Files:* Reverting to or applying the exact configuration files (hda.conf) that *worked* on the pre-6.8 kernels failed to activate the microphone on the post-6.8.0-31 kernels, suggesting a deeper structural change in the driver code. Request The failure of all standard workarounds indicates that the regression is due to a *missing or incorrect pin-mapping configuration* for the *1043:1961 SSID* within the snd-hda-intel driver, likely introduced in a patch after *6.8.0-31*. A patch is needed to re-introduce the correct microphone *quirk* for the *ASUS X509UA (1043:1961)* to restore functionality on the latest Linux kernel series.

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[PATCH REQUEST] ASUS ROG Zephyrus G14 GA403W - CS35L56 volume controls not exposed correctly
by John DelTorchio 02 Nov '25

02 Nov '25

After extensive debugging, the issue has been isolated to incorrect ALSA mixer control mapping. Technical Details Hardware Configuration: - Laptop Model: ASUS ROG Zephyrus G14 (GA403W) - Audio Codec: Realtek ALC285 (Subsystem ID: 0x10431024) - Amplifiers: Dual Cirrus Logic CS35L56 smart amplifiers - ALSA Card: Card 2 (HD-Audio Generic) - Driver: snd_hda_intel with cs35l56-hda module Amplifier Status: The CS35L56 amplifiers are being detected and initialized correctly: cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP1: Firmware: 1a00d6 vendor: 0x2 v3.11.16 cs35l56-hda i2c-CSC3556:00-cs35l56-hda.1: DSP1: Firmware: 1a00d6 vendor: 0x2 v3.11.16 Firmware files loaded: - cirrus/cs35l56-b0-dsp1-misc-10431024-spkid1.wmfw - cirrus/cs35l56-b0-dsp1-misc-10431024-spkid1-amp1.bin - cirrus/cs35l56-b0-dsp1-misc-10431024-spkid1-amp2.bin The Actual Problem Current (broken) behavior: - The volume slider controls Master Playback Volume (numid=24) on the ALC285 codec - This control only affects the analog signal level going into the CS35L56 amplifiers - Lowering this control degrades signal quality (adds muffling/distortion) without actually reducing speaker output volume - The CS35L56 amplifiers remain at a fixed gain, so perceived loudness doesn't change significantly Correct behavior should be: - The volume slider should control AMP1 Speaker Playback Volume (numid=6) and AMP2 Speaker Playback Volume (numid=12) - The codec's Master and Speaker controls should remain at 100% for optimal signal-to-noise ratio Verified working configuration: bash # Codec output (should be fixed at these values) amixer -c2 sset 'Master' 100 % amixer -c2 sset 'Speaker' 100 % # Amplifier controls (should be user-adjustable via volume slider) amixer -c2 sset 'AMP1 Speaker' 85 % amixer -c2 sset 'AMP2 Speaker' 85 % `` ` Manual testing confirms: - ✅ ` AMP1 Speaker ` and ` AMP2 Speaker ` controls properly adjust speaker loudness - ✅ Volume range 0 -100% works without distortion when codec is at 100 % - ❌ PipeWire/desktop environment uses the wrong mixer controls ### PipeWire Configuration Current sink properties show it's using ACP ( ALSA Card Profiles ) : ` ` ` api.alsa.use-acp = "true" api.alsa.path = "front:2" ` ` ` The automatically selected mixer path is choosing ` Master ` as the primary volume control instead of the CS35L56 amp controls. ### Required Fix This needs one of the following solutions: 1 . **ALSA UCM ( Use Case Manager ) profile** for device ID ` 10431024 ` : - Lock codec controls at 100 % - Expose AMP1/AMP2 Speaker controls as primary volume 2 . **Custom ALSA mixer path** at ` /usr/share/alsa/mixer/paths/ ` : - Define AMP1/AMP2 Speaker as volume elements - Set Master/Speaker to ` volume = ignore ` or ` volume-limit = 0.0 ` 3 . **Kernel quirk** in ` sound/pci/hda/patch_realtek.c ` : - Add model-specific configuration for GA403W - Similar to how other ASUS models with CS35L41/CS35L56 are handled 4 . **PipeWire/WirePlumber profile update**: - Device-specific configuration to use correct mixer elements ### Relevant Files **ALSA Controls Output:** ` `` numid = 6 ,iface = MIXER,name = 'AMP1 Speaker Playback Volume' : values = 400 | dBscale-min = -100.00dB,step = 0 .25dB,mute = 0 | Range: 0 -448 numid = 12 ,iface = MIXER,name = 'AMP2 Speaker Playback Volume' : values = 400 | dBscale-min = -100.00dB,step = 0 .25dB,mute = 0 | Range: 0 -448 numid = 24 ,iface = MIXER,name = 'Master Playback Volume' : values = 39 | dBscale-min = -65.25dB,step = 0 .75dB,mute = 0 | Range: 0 -87 Codec Information: - See full codec dump at: https://paste.centos.org/view/6c3a3640 - Full diagnostics at: https://paste.centos.org/view/c0d20bf3 Temporary Workaround Users can manually set correct levels on boot with this autostart script: ~/.config/autostart/audio-fix.desktop: desktop [Desktop Entry] Type=Application Name=Audio Fix Exec=/usr/bin/bash -c "sleep 5 && amixer -c2 sset 'Master' 100%% && amixer -c2 sset 'Speaker' 100%% && amixer -c2 sset 'AMP1 Speaker' 85%% && amixer -c2 sset 'AMP2 Speaker' 85%%" Hidden=false NoDisplay=false However, the volume slider will still not function correctly. Sent with [Proton Mail](https://proton.me/mail/home) secure email.

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doxygen: fix for doxygen 1.10.0 and later
by GitHub pull_request - opened 01 Nov '25

01 Nov '25

alsa-project/alsa-lib pull request #480 was opened from borine: doygen 1.10.0 changed the paths that include files are searched, with the consequence for alsa-lib that the "Topics" tab is no longer fully populated. See for example the [alsa project online docs](https://www.alsa-project.org/alsa-doc/alsa-lib/topics.html) To fix this we need to prevent circular #include references when doygen scans its input files, and also to re-order some macros. Request URL : https://github.com/alsa-project/alsa-lib/pull/480 Patch URL : https://github.com/alsa-project/alsa-lib/pull/480.patch Repository URL: https://github.com/alsa-project/alsa-lib

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amidirecord the metronome doesn't work
by GitHub issues - opened 31 Oct '25

31 Oct '25

alsa-project/alsa-utils issue #307 was opened from MarcWeber: This a bug report that arecordmidi's metronome doesn't work on my (nixos) linux machine for whatever reason. Asking AI to fix it I couldn't really make it work. It kept stopping after one measure. So I asked to rewrite roughly with the following prompt: - master / -slave (be metronome or use another time giving source) slave is not tested. Because goal was to get readable clean nodes I asked to add new features: --note-recording-on-switches-others-off --note-recording-ignore-off But the honest answer is using lmms like tool to record midi shows that getting timing correct is too hard. So the only way to make this work I feel would be using A UI app which allows to manually fix the timing of measures. But that would be a totally different bloated app. So typical usage of the code below would be: ``` arecordmidi --debug --master \ --note-recording-on-switches-others-off --note-recording-ignore-off \ -p 24:0 -m 128:1 -i 4:4 -t 50 /tmp/out.mid ``` But I still wasn't able to cleanly record "Alle meine Entchen" which only is made up of quarter half and full note length. The code below was done by Gemini Pro 2025-10-31 but it was fed with the original code. I also applied some manual fixes to the AI code cause it kept failing on some things. So if AI is a programmer it probably is still GPL ? No idea. So maybe test whether the metronome works for you if not add a simple patch showing this issue (for now) so that people can compile the code below themselves if they want to experimenfit with it. But they will have read the warnings here understanding it might not be as easy as they think. So I am not sure anymore the bug is worth fixing. But this issue allows discusssing it. But its worth managing expectations of users - thus drop the feature if it doesn't work or fix it by printf ("see this gituhb issue") for the time being. Maybe the code is old and was written for older Alsa no idea. The metronome output wasn't even shown in qjackctl or aconnect -l. So this all at least works in the code below so that it can be experimented with. It also outputs the pressed notes with timing differences compared to ticks. But in the end you might be faster using lilypond or musescore to type notes. ``` /* * arecordmidi.c - record standard MIDI files from sequencer * * Copyright (c) 2004-2005 Clemens Ladisch <clemens(a)ladisch.de> * Copyright (c) 2024 Rewritten by AI based on user specification. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include <stdio.h> #include <stdlib.h> #include <string.h> #include <stdarg.h> #include <signal.h> #include <getopt.h> #include <poll.h> #include <errno.h> #include <math.h> #include <time.h> /* for clock_gettime */ #include <alsa/asoundlib.h> #include "aconfig.h" #include "version.h" #define BUFFER_SIZE 4096 #define MAX_PORTS 16 /* --- Data Structures --- */ // Linked list of buffers to store SMF track data struct buffer { struct buffer *next; unsigned char buf[BUFFER_SIZE]; }; // Represents a single track in a Standard MIDI File struct smf_track { int size; int cur_buf_size; struct buffer *cur_buf; snd_seq_tick_time_t last_tick; unsigned char last_command; int used; struct buffer first_buf; }; #define TRACKS_PER_PORT 17 // 1 for meta/sysex, 16 for channels // Global application state typedef struct { enum { MODE_UNSPECIFIED, MODE_MASTER, MODE_SLAVE } mode; const char *output_filename; FILE *file; snd_seq_t *seq; int client_id; int queue_id; char *record_in_addrs[MAX_PORTS]; int record_in_count; char *metronome_out_addr; char *clock_in_addr; // Slave only char *clock_out_addr; // Master only int record_in_port_ids[MAX_PORTS]; int metronome_out_port_id; int metronome_loop_port_id; int clock_out_port_id; int bpm; int ticks_per_beat; // PPQ (Pulses Per Quarter Note) int use_metronome; int metro_channel; int metro_strong_note; int metro_weak_note; int metro_velocity; long metro_latency_ms; snd_seq_tick_time_t metro_tick_offset; int metro_running; int ts_num; int ts_den; int ts_den_pow2; int split_channels; int num_tracks; struct smf_track *tracks; volatile sig_atomic_t stop_request; int debug_mode; // New options int on_switches_others_off; int ignore_off; // For note duration calculation in debug mode snd_seq_tick_time_t active_note_start_tick[MAX_PORTS][16][128]; // For timing reference in debug mode snd_seq_tick_time_t ref_tick; snd_seq_real_time_t ref_real_time; int queue_time_ref_set; } arecordmidi_app_t; static arecordmidi_app_t app = {0}; /* --- Error Handling & Memory --- */ static void fatal(const char *msg, ...) { va_list ap; va_start(ap, msg); fputs("Error: ", stderr); vfprintf(stderr, msg, ap); fflush(stdout); va_end(ap); fputc('\n', stderr); exit(EXIT_FAILURE); } static void check_mem(void *p) { if (!p) fatal("Out of memory"); } static void check_snd(int err, const char *operation) { if (err < 0) fatal("ALSA Error: Cannot %s - %s", operation, snd_strerror(err)); } /* --- SMF File Writing Utilities --- */ static void add_byte(struct smf_track *track, unsigned char byte) { if (track->cur_buf_size >= BUFFER_SIZE) { track->cur_buf->next = calloc(1, sizeof(struct buffer)); check_mem(track->cur_buf->next); track->cur_buf = track->cur_buf->next; track->cur_buf_size = 0; } track->cur_buf->buf[track->cur_buf_size++] = byte; track->size++; } static void var_value(struct smf_track *track, unsigned int v) { unsigned long buffer = v & 0x7f; while ((v >>= 7) > 0) { buffer <<= 8; buffer |= 0x80 | (v & 0x7f); } while (1) { add_byte(track, buffer & 0xff); if (buffer & 0x80) buffer >>= 8; else break; } } static void delta_time(struct smf_track *track, const snd_seq_event_t *ev) { int diff = ev->time.tick - track->last_tick; if (diff < 0) diff = 0; var_value(track, diff); track->last_tick = ev->time.tick; } static void command(struct smf_track *track, unsigned char cmd) { if (cmd != track->last_command) { add_byte(track, cmd); track->last_command = cmd < 0xf0 ? cmd : 0; } } /* --- Metronome --- */ static void metronome_note(unsigned char note, unsigned int tick) { snd_seq_event_t ev; snd_seq_ev_clear(&ev); snd_seq_ev_set_source(&ev, app.metronome_out_port_id); snd_seq_ev_set_subs(&ev); snd_seq_ev_set_direct(&ev); unsigned int schedule_tick = tick > app.metro_tick_offset ? tick - app.metro_tick_offset : 0; if (app.debug_mode) { // printf(" OUT:METRO: Note On: ch=%2d key=%3d vel=%3d @ tick %u\n", app.metro_channel, note, app.metro_velocity, schedule_tick); } // Note On snd_seq_ev_set_noteon(&ev, app.metro_channel, note, app.metro_velocity); snd_seq_ev_schedule_tick(&ev, app.queue_id, 0, schedule_tick); // Absolute schedule snd_seq_event_output(app.seq, &ev); // Note Off (short duration) snd_seq_ev_set_noteoff(&ev, app.metro_channel, note, 0); snd_seq_ev_schedule_tick(&ev, app.queue_id, 0, schedule_tick + app.ticks_per_beat / 4); // Absolute snd_seq_event_output(app.seq, &ev); } static void metronome_pattern(unsigned int tick) { unsigned int ticks_per_bar, t; int j; ticks_per_bar = app.ts_num * (app.ticks_per_beat * 4 / app.ts_den); t = tick; for (j = 0; j < app.ts_num; ++j) { metronome_note((j == 0) ? app.metro_strong_note : app.metro_weak_note, t); t += (app.ticks_per_beat * 4 / app.ts_den); } // Schedule a custom event to trigger the next bar unsigned int next_bar_tick = tick + ticks_per_bar; snd_seq_event_t ev; snd_seq_ev_clear(&ev); ev.type = SND_SEQ_EVENT_USR0; snd_seq_ev_schedule_tick(&ev, app.queue_id, 0, next_bar_tick); // Absolute schedule snd_seq_ev_set_source(&ev, app.metronome_loop_port_id); snd_seq_ev_set_dest(&ev, app.client_id, app.metronome_loop_port_id); snd_seq_event_output(app.seq, &ev); snd_seq_drain_output(app.seq); } static void start_metronome() { if (app.metro_running || !app.use_metronome) return; app.metro_running = 1; printf("Starting metronome.\n"); // Get current tick to start metronome immediately snd_seq_queue_status_t *qstatus; snd_seq_queue_status_alloca(&qstatus); check_snd(snd_seq_get_queue_status(app.seq, app.queue_id, qstatus), "get queue status"); snd_seq_tick_time_t current_tick = snd_seq_queue_status_get_tick_time(qstatus); metronome_pattern(current_tick); } /* --- Event Recording --- */ static const char *note_names[] = {"C", "C#", "D", "D#", "E", "F", "F#", "G", "G#", "A", "A#", "B"}; static const char *midi_note_to_name(int note, char *buf, size_t size) { if (note < 0 || note > 127) { snprintf(buf, size, "???"); } else { int octave = (note / 12) - 1; const char *name = note_names[note % 12]; snprintf(buf, size, "%s%d", name, octave); } return buf; } static void terminate_old_notes(int current_port_idx, const snd_seq_event_t *original_on_ev, const snd_seq_event_t *processed_on_ev) { // If ticks_per_beat is not set, this feature is disabled to avoid division by zero. if (app.ticks_per_beat <= 0) return; unsigned int current_beat_tick = processed_on_ev->time.tick; for (int ch = 0; ch < 16; ++ch) { for (int nt = 0; nt < 128; ++nt) { snd_seq_tick_time_t start_tick = app.active_note_start_tick[current_port_idx][ch][nt]; if (start_tick != (snd_seq_tick_time_t)-1) { // Don't terminate notes started on the same tick, to allow chords. if (current_beat_tick == start_tick) { continue; } // Find the track for the note we are terminating int track_idx = current_port_idx; if (app.split_channels) { track_idx = (track_idx * TRACKS_PER_PORT) + 1 + ch; } if (track_idx >= app.num_tracks) continue; struct smf_track *track = &app.tracks[track_idx]; track->used = 1; // Create a synthetic event just to pass the correct tick time snd_seq_event_t off_ev; snd_seq_ev_clear(&off_ev); off_ev.time.tick = processed_on_ev->time.tick; // Record the NOTEOFF event delta_time(track, &off_ev); command(track, 0x80 | ch); add_byte(track, nt); add_byte(track, 0); // Standard velocity for note-off if (app.debug_mode) { char note_name_buf[8]; midi_note_to_name(nt, note_name_buf, sizeof(note_name_buf)); double beat_pos = 0.0, offset_percent = 0.0; long offset_ms = 0; if (app.ticks_per_beat > 0 && app.bpm > 0 && app.queue_time_ref_set) { beat_pos = (double)off_ev.time.tick / app.ticks_per_beat; snd_seq_tick_time_t nearest_beat_tick = floor(beat_pos + 0.5) * app.ticks_per_beat; if (snd_seq_ev_is_real(original_on_ev)) { // Precise timing: Compare event real time to ideal grid real time double tick_ns = (double)app.bpm * app.ticks_per_beat; double ns_per_tick = (60.0 * 1000000000.0) / tick_ns; long long ref_ns = (long long)app.ref_real_time.tv_sec * 1000000000LL + app.ref_real_time.tv_nsec; long long ticks_from_ref_to_beat = (long long)nearest_beat_tick - (long long)app.ref_tick; long long expected_beat_ns = ref_ns + (long long)(ticks_from_ref_to_beat * ns_per_tick); long long actual_event_ns = (long long)original_on_ev->time.time.tv_sec * 1000000000LL + original_on_ev->time.time.tv_nsec; long long ns_offset = actual_event_ns - expected_beat_ns; if (ns_offset > tick_ns / 2 ) ns_offset = ns_offset - tick_ns; offset_ms = ns_offset / 1000000LL; double ns_per_beat = ns_per_tick * app.ticks_per_beat; if (ns_per_beat > 0) offset_percent = (double)ns_offset * 100.0 / ns_per_beat; } else { // Fallback to tick-based timing for events that came from the queue long tick_offset = off_ev.time.tick - nearest_beat_tick; double ms_per_tick = (60.0 * 1000.0) / ((double)app.bpm * app.ticks_per_beat); offset_ms = round((double)tick_offset * ms_per_tick); offset_percent = (double)tick_offset * 100.0 / app.ticks_per_beat; } } if (offset_percent > 50) offset_percent = offset_percent - 100; snd_seq_tick_time_t duration_ticks = off_ev.time.tick - start_tick; double duration_beats = (app.ticks_per_beat > 0) ? (double)duration_ticks / app.ticks_per_beat : 0.0; printf("DEBUG: OFF %-4s ch=%2d vel=%3d @ beat %.2f | timing: %+.2f%% (%+ldms) | duration: %.2f beats | auto-terminated by new note \n", note_name_buf, ch, 0, beat_pos, offset_percent, offset_ms, duration_beats); } // Update state to mark note as off app.active_note_start_tick[current_port_idx][ch][nt] = (snd_seq_tick_time_t)-1; } } } } static void record_event(const snd_seq_event_t *ev) { int port_idx, track_idx; struct smf_track *track; snd_seq_event_t event_with_tick; // A mutable copy of the event with a guaranteed tick timestamp // Filter 1: Must be from our recording queue. // We accept both tick and real-time events. The ALSA sequencer is expected // to deliver a tick-stamped event from the queue, but in practice, it // sometimes only delivers the initial real-time event. if (ev->queue != app.queue_id) { if (app.debug_mode && ev->type >= SND_SEQ_EVENT_NOTEON && ev->type <= SND_SEQ_EVENT_SYSEX) { // printf(" -> REJECTED (Filter 1): Event from queue %d is not our recording queue %d.\n", ev->queue, app.queue_id); } return; } // Create a mutable copy of the event and normalize its timestamp to ticks. event_with_tick = *ev; if (snd_seq_ev_is_tick(ev)) { // Event already has a tick timestamp, use it as-is. // This is the case for metronome loopback and correctly scheduled MIDI events. } else if (snd_seq_ev_is_real(ev)) { // This is a "raw" real-time event. We convert its timestamp to a tick // value by querying the current state of our recording queue. snd_seq_queue_status_t *qstatus; snd_seq_queue_status_alloca(&qstatus); if (snd_seq_get_queue_status(app.seq, app.queue_id, qstatus) < 0) { fprintf(stderr, "Warning: could not get queue status to timestamp a real-time event; event ignored.\n"); fflush(stdout); return; } event_with_tick.time.tick = snd_seq_queue_status_get_tick_time(qstatus); if (app.debug_mode) { // printf(" -> INFO: Handled real-time event by converting to tick %u\n", event_with_tick.time.tick); } } else { // Event has no timestamp; it cannot be recorded. return; } // Filter 2: Must be destined for one of our recording input ports port_idx = -1; for (int i = 0; i < app.record_in_count; ++i) { if (event_with_tick.dest.port == app.record_in_port_ids[i]) { port_idx = i; break; } } if (port_idx == -1) { if (app.debug_mode && event_with_tick.type >= SND_SEQ_EVENT_NOTEON && event_with_tick.type <= SND_SEQ_EVENT_SYSEX) { // printf(" -> REJECTED (Filter 2): Destination port %d is not on the record list.\n", event_with_tick.dest.port); } return; } track_idx = port_idx; if (app.split_channels) { track_idx *= TRACKS_PER_PORT; if (snd_seq_ev_is_channel_type(&event_with_tick)) track_idx += 1 + (event_with_tick.data.note.channel & 0xf); } if (track_idx >= app.num_tracks) return; track = &app.tracks[track_idx]; track->used = 1; // From here on, we use our normalized `event_with_tick` for all processing. switch (event_with_tick.type) { case SND_SEQ_EVENT_NOTEON: if (event_with_tick.data.note.velocity == 0) goto handle_noteoff; // Running status note-off if (app.on_switches_others_off) { terminate_old_notes(port_idx, ev, &event_with_tick); } app.active_note_start_tick[port_idx][event_with_tick.data.note.channel & 0xf][event_with_tick.data.note.note & 0x7f] = event_with_tick.time.tick; if (app.debug_mode) { char note_name_buf[8]; midi_note_to_name(event_with_tick.data.note.note, note_name_buf, sizeof(note_name_buf)); double beat_pos = 0.0, offset_percent = 0.0; long offset_ms = 0; if (app.ticks_per_beat > 0 && app.bpm > 0 && app.queue_time_ref_set) { beat_pos = (double)event_with_tick.time.tick / app.ticks_per_beat; snd_seq_tick_time_t nearest_beat_tick = floor(beat_pos + 0.5) * app.ticks_per_beat; if (snd_seq_ev_is_real(ev)) { // Precise timing: Compare event real time to ideal grid real time double ns_per_tick = (60.0 * 1000000000.0) / ((double)app.bpm * app.ticks_per_beat); long long ref_ns = (long long)app.ref_real_time.tv_sec * 1000000000LL + app.ref_real_time.tv_nsec; long long ticks_from_ref_to_beat = (long long)nearest_beat_tick - (long long)app.ref_tick; long long expected_beat_ns = ref_ns + (long long)(ticks_from_ref_to_beat * ns_per_tick); long long actual_event_ns = (long long)ev->time.time.tv_sec * 1000000000LL + ev->time.time.tv_nsec; long long ns_offset = actual_event_ns - expected_beat_ns; if (ns_offset > ns_per_tick / 2) ns_offset -= ns_per_tick; offset_ms = ns_offset / 1000000LL; offset_percent = (double)ns_offset * 100.0 / ns_per_tick; } else { // Fallback to tick-based timing for events that came from the queue long tick_offset = event_with_tick.time.tick - nearest_beat_tick; double ms_per_tick = (60.0 * 1000.0) / ((double)app.bpm * app.ticks_per_beat); offset_ms = round((double)tick_offset * ms_per_tick); if (offset_ms > ms_per_tick) offset_ms -= ms_per_tick; offset_percent = offset_ms / ms_per_tick; } } printf("DEBUG: ON %-4s ch=%2d vel=%3d @ beat %.2f | timing: %+.2f%% (%+ldms)\n", note_name_buf, event_with_tick.data.note.channel, event_with_tick.data.note.velocity, beat_pos, offset_percent, offset_ms); fflush(stdout); } delta_time(track, &event_with_tick); command(track, 0x90 | (event_with_tick.data.note.channel & 0xf)); add_byte(track, event_with_tick.data.note.note & 0x7f); add_byte(track, event_with_tick.data.note.velocity & 0x7f); break; case SND_SEQ_EVENT_NOTEOFF: handle_noteoff: if (app.ignore_off) return; if (app.debug_mode) { unsigned char channel = event_with_tick.data.note.channel & 0xf; unsigned char note = event_with_tick.data.note.note & 0x7f; snd_seq_tick_time_t start_tick = app.active_note_start_tick[port_idx][channel][note]; char note_name_buf[8]; midi_note_to_name(note, note_name_buf, sizeof(note_name_buf)); double beat_pos = 0.0, offset_percent = 0.0; long offset_ms = 0; if (app.ticks_per_beat > 0 && app.bpm > 0 && app.queue_time_ref_set) { beat_pos = (double)event_with_tick.time.tick / app.ticks_per_beat; snd_seq_tick_time_t nearest_beat_tick = floor(beat_pos + 0.5) * app.ticks_per_beat; if (snd_seq_ev_is_real(ev)) { // Precise timing: Compare event real time to ideal grid real time double ns_per_tick = (60.0 * 1000000000.0) / ((double)app.bpm * app.ticks_per_beat); long long ref_ns = (long long)app.ref_real_time.tv_sec * 1000000000LL + app.ref_real_time.tv_nsec; long long ticks_from_ref_to_beat = (long long)nearest_beat_tick - (long long)app.ref_tick; long long expected_beat_ns = ref_ns + (long long)(ticks_from_ref_to_beat * ns_per_tick); long long actual_event_ns = (long long)ev->time.time.tv_sec * 1000000000LL + ev->time.time.tv_nsec; long long ns_offset = actual_event_ns - expected_beat_ns; offset_ms = ns_offset / 1000000LL; double ns_per_beat = ns_per_tick * app.ticks_per_beat; if (ns_per_beat > 0) offset_percent = (double)ns_offset * 100.0 / ns_per_beat; } else { // Fallback to tick-based timing for events that came from the queue long tick_offset = event_with_tick.time.tick - nearest_beat_tick; double ms_per_tick = (60.0 * 1000.0) / ((double)app.bpm * app.ticks_per_beat); offset_ms = round((double)tick_offset * ms_per_tick); offset_percent = (double)tick_offset * 100.0 / app.ticks_per_beat; } } snd_seq_tick_time_t duration_ticks = 0; if (start_tick != (snd_seq_tick_time_t)-1 && event_with_tick.time.tick >= start_tick) { duration_ticks = event_with_tick.time.tick - start_tick; } double duration_beats = (app.ticks_per_beat > 0) ? (double)duration_ticks / app.ticks_per_beat : 0.0; printf("DEBUG: OFF %-4s ch=%2d vel=%3d @ beat %.2f | timing: %+.2f%% (%+ldms) | duration: %.2f beats\n", note_name_buf, event_with_tick.data.note.channel, event_with_tick.data.note.velocity, beat_pos, offset_percent, offset_ms, duration_beats); fflush(stdout); } app.active_note_start_tick[port_idx][event_with_tick.data.note.channel & 0xf][event_with_tick.data.note.note & 0x7f] = (snd_seq_tick_time_t)-1; delta_time(track, &event_with_tick); command(track, 0x80 | (event_with_tick.data.note.channel & 0xf)); add_byte(track, event_with_tick.data.note.note & 0x7f); add_byte(track, event_with_tick.data.note.velocity & 0x7f); break; case SND_SEQ_EVENT_PGMCHANGE: delta_time(track, &event_with_tick); command(track, 0xc0 | (event_with_tick.data.control.channel & 0xf)); add_byte(track, event_with_tick.data.control.value & 0x7f); break; case SND_SEQ_EVENT_CONTROLLER: delta_time(track, &event_with_tick); command(track, 0xb0 | (event_with_tick.data.control.channel & 0xf)); add_byte(track, event_with_tick.data.control.param & 0x7f); add_byte(track, event_with_tick.data.control.value & 0x7f); break; case SND_SEQ_EVENT_PITCHBEND: delta_time(track, &event_with_tick); command(track, 0xe0 | (event_with_tick.data.control.channel & 0xf)); add_byte(track, (event_with_tick.data.control.value + 8192) & 0x7f); add_byte(track, ((event_with_tick.data.control.value + 8192) >> 7) & 0x7f); break; case SND_SEQ_EVENT_SYSEX: delta_time(track, &event_with_tick); add_byte(track, 0xf0); track->last_command = 0; var_value(track, event_with_tick.data.ext.len); for (unsigned int i = 0; i < event_with_tick.data.ext.len; ++i) add_byte(track, ((unsigned char*)event_with_tick.data.ext.ptr)[i]); break; default: break; } } /* --- Setup and Teardown --- */ static void setup_tracks() { app.num_tracks = app.record_in_count; if (app.split_channels) app.num_tracks *= TRACKS_PER_PORT; app.tracks = calloc(app.num_tracks, sizeof(struct smf_track)); check_mem(app.tracks); for (int i = 0; i < app.num_tracks; ++i) app.tracks[i].cur_buf = &app.tracks[i].first_buf; app.tracks[0].used = 1; // Track 0 always used for meta events } static void write_initial_meta_events() { // Tempo int usecs_per_quarter = 60000000 / app.bpm; var_value(&app.tracks[0], 0); add_byte(&app.tracks[0], 0xff); add_byte(&app.tracks[0], 0x51); add_byte(&app.tracks[0], 3); add_byte(&app.tracks[0], usecs_per_quarter >> 16); add_byte(&app.tracks[0], (usecs_per_quarter >> 8) & 0xff); add_byte(&app.tracks[0], usecs_per_quarter & 0xff); // Time Signature var_value(&app.tracks[0], 0); add_byte(&app.tracks[0], 0xff); add_byte(&app.tracks[0], 0x58); add_byte(&app.tracks[0], 4); add_byte(&app.tracks[0], app.ts_num); add_byte(&app.tracks[0], app.ts_den_pow2); add_byte(&app.tracks[0], 24); // MIDI clocks per metronome click add_byte(&app.tracks[0], 8); // 32nd notes per quarter note } static void terminate_all_active_notes(void) { if (!app.ignore_off) return; snd_seq_queue_status_t *qstatus; snd_seq_queue_status_alloca(&qstatus); check_snd(snd_seq_get_queue_status(app.seq, app.queue_id, qstatus), "get queue status"); snd_seq_tick_time_t final_tick = snd_seq_queue_status_get_tick_time(qstatus); // printf("Terminating any remaining notes at tick %u...\n", final_tick); for (int p_idx = 0; p_idx < app.record_in_count; ++p_idx) { for (int ch = 0; ch < 16; ++ch) { for (int nt = 0; nt < 128; ++nt) { snd_seq_tick_time_t start_tick = app.active_note_start_tick[p_idx][ch][nt]; if (start_tick != (snd_seq_tick_time_t)-1) { int track_idx = p_idx; if (app.split_channels) { track_idx = (p_idx * TRACKS_PER_PORT) + 1 + ch; } if (track_idx >= app.num_tracks) continue; struct smf_track *track = &app.tracks[track_idx]; track->used = 1; // Mark track as used if it wasn't already snd_seq_event_t off_ev; snd_seq_ev_clear(&off_ev); off_ev.time.tick = final_tick; delta_time(track, &off_ev); command(track, 0x80 | ch); add_byte(track, nt); add_byte(track, 0); if (app.debug_mode) { char note_name_buf[8]; midi_note_to_name(nt, note_name_buf, sizeof(note_name_buf)); double beat_pos = 0.0, offset_percent = 0.0; long offset_ms = 0; if (app.ticks_per_beat > 0 && app.bpm > 0 && app.queue_time_ref_set) { beat_pos = (double)off_ev.time.tick / app.ticks_per_beat; // NOTE: Cannot calculate real-time offset as we don't have a real event timestamp. // We fall back to calculating offset from the nearest beat based on ticks alone. long nearest_beat_tick = floor(beat_pos + 0.5) * app.ticks_per_beat; long tick_offset = off_ev.time.tick - nearest_beat_tick; double ms_per_tick = (60.0 * 1000.0) / ((double)app.bpm * app.ticks_per_beat); offset_ms = round((double)tick_offset * ms_per_tick); if (offset_ms > ms_per_tick/ 2) offset_ms -= ms_per_tick; offset_percent = offset_ms / ms_per_tick; } snd_seq_tick_time_t duration_ticks = off_ev.time.tick - start_tick; double duration_beats = (app.ticks_per_beat > 0) ? (double)duration_ticks / app.ticks_per_beat : 0.0; printf("DEBUG: OFF %-4s ch=%2d vel=%3d @ beat %.2f | auto-terminated at exit | timing: %+.2f%% (%+ldms) | duration: %.2f beats\n", note_name_buf, ch, 0, beat_pos, offset_percent, offset_ms, duration_beats); fflush(stdout); } } } } } } static void finish_tracks() { snd_seq_queue_status_t *qstatus; snd_seq_queue_status_alloca(&qstatus); check_snd(snd_seq_get_queue_status(app.seq, app.queue_id, qstatus), "get queue status"); snd_seq_tick_time_t tick = snd_seq_queue_status_get_tick_time(qstatus); for (int i = 0; i < app.num_tracks; ++i) { if (!app.tracks[i].used) continue; int diff = tick - app.tracks[i].last_tick; if (diff < 0) diff = 0; var_value(&app.tracks[i], diff); add_byte(&app.tracks[i], 0xff); add_byte(&app.tracks[i], 0x2f); add_byte(&app.tracks[i], 0); } } static void write_smf_file() { int used_tracks = 0, i; for (i = 0; i < app.num_tracks; ++i) if (app.tracks[i].used) used_tracks++; if (used_tracks == 0) { printf("No events recorded, output file will not be created.\n"); return; } app.file = fopen(app.output_filename, "wb"); if (!app.file) fatal("Cannot open %s for writing - %s", app.output_filename, strerror(errno)); // MThd chunk fwrite("MThd\0\0\0\6", 1, 8, app.file); fputc(0, app.file); fputc(used_tracks > 1 ? 1 : 0, app.file); fputc((used_tracks >> 8) & 0xff, app.file); fputc(used_tracks & 0xff, app.file); fputc(app.ticks_per_beat >> 8, app.file); fputc(app.ticks_per_beat & 0xff, app.file); // MTrk chunks for (i = 0; i < app.num_tracks; ++i) { struct smf_track *track = &app.tracks[i]; if (!track->used) continue; fwrite("MTrk", 1, 4, app.file); fputc((track->size >> 24) & 0xff, app.file); fputc((track->size >> 16) & 0xff, app.file); fputc((track->size >> 8) & 0xff, app.file); fputc(track->size & 0xff, app.file); struct buffer *buf; for (buf = &track->first_buf; buf; buf = buf->next) fwrite(buf->buf, 1, (buf == track->cur_buf) ? track->cur_buf_size : BUFFER_SIZE, app.file); } fclose(app.file); } static void setup_alsa() { snd_seq_port_info_t *pinfo; snd_seq_addr_t addr; check_snd(snd_seq_open(&app.seq, "default", SND_SEQ_OPEN_DUPLEX, 0), "open sequencer"); app.client_id = snd_seq_client_id(app.seq); check_snd(app.client_id, "get client id"); check_snd(snd_seq_set_client_name(app.seq, "arecordmidi"), "set client name"); app.queue_id = snd_seq_alloc_named_queue(app.seq, "arecordmidi queue"); check_snd(app.queue_id, "create queue"); snd_seq_port_info_alloca(&pinfo); for (int i = 0; i < app.record_in_count; ++i) { char port_name[64]; snprintf(port_name, sizeof(port_name), "Record In %d", i + 1); app.record_in_port_ids[i] = snd_seq_create_simple_port(app.seq, port_name, SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE, SND_SEQ_PORT_TYPE_APPLICATION); check_snd(app.record_in_port_ids[i], "create record port"); // Configure the port for timestamping events and scheduling them on our queue check_snd(snd_seq_get_port_info(app.seq, app.record_in_port_ids[i], pinfo), "get port info"); snd_seq_port_info_set_timestamping(pinfo, 1); snd_seq_port_info_set_timestamp_real(pinfo, 1); snd_seq_port_info_set_timestamp_queue(pinfo, app.queue_id); check_snd(snd_seq_set_port_info(app.seq, app.record_in_port_ids[i], pinfo), "configure record port"); // Connect the external source to our new port using an explicit subscription // that specifies the queue. This is more robust than relying on the port's // properties alone. snd_seq_port_subscribe_t *subs; snd_seq_port_subscribe_alloca(&subs); check_snd(snd_seq_parse_address(app.seq, &addr, app.record_in_addrs[i]), app.record_in_addrs[i]); snd_seq_addr_t my_dest_addr; my_dest_addr.client = app.client_id; my_dest_addr.port = app.record_in_port_ids[i]; snd_seq_port_subscribe_set_sender(subs, &addr); snd_seq_port_subscribe_set_dest(subs, &my_dest_addr); snd_seq_port_subscribe_set_queue(subs, app.queue_id); check_snd(snd_seq_subscribe_port(app.seq, subs), "connect record port"); } if (app.use_metronome) { app.metronome_out_port_id = snd_seq_create_simple_port(app.seq, "Metronome Out", SND_SEQ_PORT_CAP_READ | SND_SEQ_PORT_CAP_SUBS_READ, SND_SEQ_PORT_TYPE_APPLICATION); check_snd(app.metronome_out_port_id, "create metronome out port"); check_snd(snd_seq_parse_address(app.seq, &addr, app.metronome_out_addr), app.metronome_out_addr); check_snd(snd_seq_connect_to(app.seq, app.metronome_out_port_id, addr.client, addr.port), "connect metronome port"); app.metronome_loop_port_id = snd_seq_create_simple_port(app.seq, "Metronome Loop", SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE, SND_SEQ_PORT_TYPE_APPLICATION); check_snd(app.metronome_loop_port_id, "create metronome loop port"); check_snd(snd_seq_connect_from(app.seq, app.metronome_loop_port_id, app.client_id, app.metronome_loop_port_id), "connect metronome loopback"); } if (app.mode == MODE_MASTER) { snd_seq_queue_tempo_t *tempo; snd_seq_queue_tempo_alloca(&tempo); snd_seq_queue_tempo_set_tempo(tempo, 60000000 / app.bpm); snd_seq_queue_tempo_set_ppq(tempo, app.ticks_per_beat); check_snd(snd_seq_set_queue_tempo(app.seq, app.queue_id, tempo), "set queue tempo"); app.clock_out_port_id = snd_seq_create_simple_port(app.seq, "Clock Out", SND_SEQ_PORT_CAP_READ | SND_SEQ_PORT_CAP_SUBS_READ, SND_SEQ_PORT_TYPE_MIDI_GENERIC); check_snd(app.clock_out_port_id, "create clock out port"); snd_seq_addr_t sender = { .client = SND_SEQ_CLIENT_SYSTEM, .port = SND_SEQ_PORT_SYSTEM_TIMER }; snd_seq_addr_t dest = { .client = app.client_id, .port = app.clock_out_port_id }; snd_seq_port_subscribe_t *subs; snd_seq_port_subscribe_alloca(&subs); snd_seq_port_subscribe_set_sender(subs, &sender); snd_seq_port_subscribe_set_dest(subs, &dest); snd_seq_port_subscribe_set_queue(subs, app.queue_id); snd_seq_port_subscribe_set_time_update(subs, 1); check_snd(snd_seq_subscribe_port(app.seq, subs), "subscribe clock out to system timer"); if (app.clock_out_addr) { check_snd(snd_seq_parse_address(app.seq, &addr, app.clock_out_addr), app.clock_out_addr); check_snd(snd_seq_connect_to(app.seq, app.clock_out_port_id, addr.client, addr.port), "connect clock out port"); } } else { // MODE_SLAVE snd_seq_queue_timer_t *timer; snd_seq_queue_timer_alloca(&timer); snd_seq_queue_timer_set_type(timer, SND_SEQ_TIMER_ALSA); snd_seq_queue_timer_set_id(timer, 0); check_snd(snd_seq_set_queue_timer(app.seq, app.queue_id, timer), "set queue timer"); snd_seq_addr_t sender, dest; check_snd(snd_seq_parse_address(app.seq, &sender, app.clock_in_addr), app.clock_in_addr); dest.client = SND_SEQ_CLIENT_SYSTEM; dest.port = SND_SEQ_PORT_SYSTEM_TIMER; snd_seq_port_subscribe_t *subs; snd_seq_port_subscribe_alloca(&subs); snd_seq_port_subscribe_set_sender(subs, &sender); snd_seq_port_subscribe_set_dest(subs, &dest); check_snd(snd_seq_subscribe_port(app.seq, subs), "subscribe system timer to external clock"); } } static void list_ports_and_exit() { snd_seq_client_info_t *cinfo; snd_seq_port_info_t *pinfo; snd_seq_t *seq; check_snd(snd_seq_open(&seq, "default", SND_SEQ_OPEN_INPUT, 0), "open sequencer for listing"); snd_seq_client_info_alloca(&cinfo); snd_seq_port_info_alloca(&pinfo); puts(" Port Client name Port name"); puts("-----------------------------------------------------------------"); snd_seq_client_info_set_client(cinfo, -1); while (snd_seq_query_next_client(seq, cinfo) >= 0) { int client = snd_seq_client_info_get_client(cinfo); snd_seq_port_info_set_client(pinfo, client); snd_seq_port_info_set_port(pinfo, -1); while (snd_seq_query_next_port(seq, pinfo) >= 0) { unsigned int caps = snd_seq_port_info_get_capability(pinfo); if ((caps & SND_SEQ_PORT_CAP_SUBS_READ) || (caps & SND_SEQ_PORT_CAP_READ)) { printf("%3d:%-3d %-32.32s %s\n", snd_seq_port_info_get_client(pinfo), snd_seq_port_info_get_port(pinfo), snd_seq_client_info_get_name(cinfo), snd_seq_port_info_get_name(pinfo)); fflush(stdout); } } } snd_seq_close(seq); exit(0); } /* --- Main Logic --- */ static void sighandler(int signal) { (void)signal; app.stop_request = 1; } static void help(const char *argv0) { fprintf(stderr, "Usage: %s [options] outputfile.mid\n" "\nMODES (choose one, --master is default):\n" " --master Provide MIDI clock and be the master (default)\n" " --slave Receive MIDI clock from another device\n" "\nPORT CONFIGURATION:\n" " -p, --midi-in-record=c:p Add a source port to record from (can be used multiple times)\n" " -m, --midi-out-metronome=c:p Play metronome to this destination port\n" " --midi-in-clock=c:p In slave mode, receive clock from here\n" " --midi-out-clock=c:p In master mode, send MIDI clock to here (optional)\n" "\nTIMING & RECORDING:\n" " -b, --bpm=beats Tempo in beats per minute (master mode)\n" " -t, --ticks=ticks Resolution in ticks per beat (PPQ)\n" " -i, --timesig=nn:dd Time signature for metronome (e.g., 4:4, 3:4)\n" " -s, --split-channels Create a separate track for each MIDI channel\n" " --note-recording-on-switches-others-off\n" " A new note-on event terminates other sounding notes\n" " --note-recording-ignore-off\n" " Ignore note-off events (implies the above option)\n" "\nMETRONOME TWEAKS:\n" " --compensate-metronome-output-latency-ms=<ms> Send metronome notes early\n" "\nOTHER OPTIONS:\n" " -h, --help This help\n" " -V, --version Show version\n" " -l, --list List available MIDI ports\n" " --debug Output recorded events to the console\n", argv0); } static void parse_time_signature(const char *arg) { char *sep; app.ts_num = strtol(arg, &sep, 10); if (app.ts_num < 1 || app.ts_num > 64 || (*sep != ':' && *sep != '/')) fatal("Invalid time signature numerator: %s", arg); app.ts_den = strtol(++sep, NULL, 10); if (app.ts_den < 1 || (app.ts_den & (app.ts_den - 1)) != 0) // Must be power of 2 fatal("Invalid time signature denominator (must be power of 2): %s", arg); app.ts_den_pow2 = 0; for (int x = app.ts_den; x > 1; x /= 2) ++app.ts_den_pow2; } static void validate_args() { if (app.mode == MODE_UNSPECIFIED) app.mode = MODE_MASTER; if (app.record_in_count == 0) fatal("Please specify at least one source port with -p or --midi-in-record."); if (app.mode == MODE_SLAVE && !app.clock_in_addr) fatal("Slave mode requires --midi-in-clock."); if (app.mode == MODE_MASTER && app.clock_in_addr) fatal("--midi-in-clock can only be used in slave mode."); if (app.mode == MODE_SLAVE && app.clock_out_addr) fatal("--midi-out-clock can only be used in master mode."); if (app.use_metronome && !app.metronome_out_addr) fatal("Using metronome requires specifying --midi-out-metronome."); if (app.metro_latency_ms > 0) { double ticks_per_ms = (app.bpm / 60.0 * app.ticks_per_beat) / 1000.0; app.metro_tick_offset = (snd_seq_tick_time_t)(app.metro_latency_ms * ticks_per_ms); } if (app.ignore_off) { app.on_switches_others_off = 1; } } int main(int argc, char *argv[]) { app.mode = MODE_UNSPECIFIED; app.bpm = 120; app.ticks_per_beat = 384; app.ts_num = 4; app.ts_den = 4; app.ts_den_pow2 = 2; // Default 4/4 app.metro_channel = 9; app.metro_strong_note = 34; app.metro_weak_note = 33; app.metro_velocity = 100; app.queue_time_ref_set = 0; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, {"list", 0, NULL, 'l'}, {"master", 0, NULL, 1000}, {"slave", 0, NULL, 1001}, {"midi-in-record", 1, NULL, 'p'}, {"midi-out-metronome", 1, NULL, 'm'}, {"midi-in-clock", 1, NULL, 1002}, {"midi-out-clock", 1, NULL, 1003}, {"bpm", 1, NULL, 'b'}, {"ticks", 1, NULL, 't'}, {"timesig", 1, NULL, 'i'}, {"split-channels", 0, NULL, 's'}, {"debug", 0, NULL, 1004}, {"compensate-metronome-output-latency-ms", 1, NULL, 1005}, {"note-recording-on-switches-others-off", 0, NULL, 1006}, {"note-recording-ignore-off", 0, NULL, 1007}, {0} }; int c; while ((c = getopt_long(argc, argv, "hVlp:m:b:t:i:s", long_options, NULL)) != -1) { switch (c) { case 'h': help(argv[0]); return 0; case 'V': fputs("arecordmidi version " SND_UTIL_VERSION_STR "\n", stderr); return 0; case 'l': list_ports_and_exit(); break; case 1000: if (app.mode != MODE_UNSPECIFIED) fatal("--master and --slave are mutually exclusive."); app.mode = MODE_MASTER; break; case 1001: if (app.mode != MODE_UNSPECIFIED) fatal("--master and --slave are mutually exclusive."); app.mode = MODE_SLAVE; break; case 'p': if (app.record_in_count >= MAX_PORTS) fatal("Too many --midi-in-record ports (max %d)", MAX_PORTS); app.record_in_addrs[app.record_in_count++] = optarg; break; case 'm': app.metronome_out_addr = optarg; app.use_metronome = 1; break; case 1002: app.clock_in_addr = optarg; break; case 1003: app.clock_out_addr = optarg; break; case 'b': app.bpm = atoi(optarg); break; case 't': app.ticks_per_beat = atoi(optarg); break; case 'i': parse_time_signature(optarg); break; case 's': app.split_channels = 1; break; case 1004: app.debug_mode = 1; break; case 1005: app.metro_latency_ms = atol(optarg); break; case 1006: app.on_switches_others_off = 1; break; case 1007: app.ignore_off = 1; break; default: help(argv[0]); return 1; } } if (optind >= argc) fatal("Please specify an output file name."); app.output_filename = argv[optind]; validate_args(); // Initialize the active note tracker for debug output memset(app.active_note_start_tick, -1, sizeof(app.active_note_start_tick)); setup_alsa(); setup_tracks(); write_initial_meta_events(); if (app.mode == MODE_MASTER) { printf("Master mode: Starting recording at %d BPM.\n", app.bpm); check_snd(snd_seq_start_queue(app.seq, app.queue_id, NULL), "start queue"); // For debug timing, get a reliable reference time from ALSA itself // immediately after starting the queue. This avoids clock source mismatches. if (app.debug_mode) { snd_seq_queue_status_t *qstatus; snd_seq_queue_status_alloca(&qstatus); check_snd(snd_seq_get_queue_status(app.seq, app.queue_id, qstatus), "get queue status for time ref"); app.ref_tick = snd_seq_queue_status_get_tick_time(qstatus); app.ref_real_time = *snd_seq_queue_status_get_real_time(qstatus); app.queue_time_ref_set = 1; } start_metronome(); } else { printf("Slave mode: Waiting for MIDI clock from %s...\n", app.clock_in_addr); snd_seq_control_queue(app.seq, app.queue_id, SND_SEQ_EVENT_START, 0, NULL); } signal(SIGINT, sighandler); signal(SIGTERM, sighandler); printf("Recording to '%s'. Press Ctrl+C to stop.\n", app.output_filename); int npfds = snd_seq_poll_descriptors_count(app.seq, POLLIN); struct pollfd *pfds = alloca(sizeof(*pfds) * npfds); while (!app.stop_request) { snd_seq_poll_descriptors(app.seq, pfds, npfds, POLLIN); if (poll(pfds, npfds, 1000) < 0 && errno != EINTR) break; snd_seq_event_t *ev; while (snd_seq_event_input(app.seq, &ev) > 0 && !app.stop_request) { if (app.debug_mode) { // printf("INPUT EVENT: type=%-3d src=%-3d:%-3d dest=%-3d:%-3d queue=%-3d ", // ev->type, // ev->source.client, ev->source.port, // ev->dest.client, ev->dest.port, // ev->queue); if (snd_seq_ev_is_tick(ev)) { // printf("time=tick:%u\n", ev->time.tick); } else if (snd_seq_ev_is_real(ev)) { // printf("time=real:%ld.%09ld\n", (long)ev->time.time.tv_sec, (long)ev->time.time.tv_nsec); } else { // printf("time=none\n"); } } // Handle slave mode start, which starts the queue if (ev->type == SND_SEQ_EVENT_START && app.mode == MODE_SLAVE) { printf("Received MIDI Start. Starting recording and metronome.\n"); if (app.debug_mode && !app.queue_time_ref_set) { app.ref_tick = ev->time.tick; // should be 0 app.ref_real_time = ev->time.time; app.queue_time_ref_set = 1; } start_metronome(); } // Handle metronome loopback event if (app.use_metronome && ev->dest.port == app.metronome_loop_port_id && ev->type == SND_SEQ_EVENT_USR0) { metronome_pattern(ev->time.tick); } // Attempt to record any other event. The function itself will filter. record_event(ev); snd_seq_free_event(ev); } } printf("\nRecording stopped. Finalizing file...\n"); terminate_all_active_notes(); finish_tracks(); write_smf_file(); for (int i = 0; i < app.num_tracks; ++i) { struct buffer *b = app.tracks[i].first_buf.next, *next; while (b) { next = b->next; free(b); b = next; } } free(app.tracks); snd_seq_close(app.seq); printf("Done.\n"); return 0; } ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/307 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

ucm2: sof: hdmi: Add 5.1 and 7.1 variants of HDMI devices when using IPC4
by GitHub pull_request - edited 31 Oct '25

31 Oct '25

alsa-project/alsa-ucm-conf pull request #633 was edited from ujfalusi: Systems using IPC4 can support up to 8 channels of audio (and passthrough) via HDMI. In UCM the default PlaybackChannels is set to 2, which prevents users from selecting multichannel configurations. When probing the card, Pipewire will drop configurations that are not supported either by the PCM device or based on ELD information. This means that if the equipment supports only stereo then the 5.1 and 7.1 variants should not be visible, if the equipment is 5.1 capable, then only the 7.1 variant is removed. The kernel will refine the PCM parameters based on the ELD information as wall when https://lore.kernel.org/linux-sound/20251029073600.13624-1-peter.ujfalusi@l… is applied. @perexg, @wtay, @ford-prefect, is this something which can help PW and user space to handle the HDMI a bit better with SOF? Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/633 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/633.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof: hdmi: Add 5.1 and 7.1 varriants of HDMI devices when using IPC4
by GitHub pull_request - opened 31 Oct '25

31 Oct '25

alsa-project/alsa-ucm-conf pull request #633 was opened from ujfalusi: Systems using IPC4 can support up to 8 channels of audio (and passthrough) via HDMI. In UCM the default PlaybackChannels is set to 2, which prevents users from selecting multichannel configurations. When probing the card, Pipewire will drop configurations that are not supported either by the PCM device or based on ELD information. This means that if the equipment supports only stereo then the 5.1 and 7.1 variants should not be visible, if the equipment is 5.1 capable, then only the 7.1 variant is removed. The kernel will refine the PCM parameters based on the ELD information as wall when https://lore.kernel.org/linux-sound/20251029073600.13624-1-peter.ujfalusi@l… is applied. @perexg, @wtay, @ford-prefect, is this something which can help PW and user space to handle the HDMI a bit better with SOF? Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/633 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/633.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Raptor Lake external jack needs manual ALSA unmute and speakers don’t auto-mute
by GitHub issues - opened 31 Oct '25

31 Oct '25

alsa-project/alsa-ucm-conf issue #632 was opened from keponk: SOF HDA (skl_hda_dsp_generic) on Raptor Lake: Headphone/Line-out not exposed as a separate port to PipeWire; external jack needs manual ALSA unmute and speakers don’t auto-mute Environment (for quick identification) * Distro: NixOS (current channel) * Audio stack: PipeWire/WirePlumber 1.4.7, PipeWire Pulse * Hardware: Intel Raptor Lake PCH HDA (onboard), SOF HDA path (skl_hda_dsp_generic) * Symptom device string seen in userspace: “Raptor Lake High Definition Audio Controller” * PipeWire sink name: alsa_output.pci-0000_00_1f.3-platform-skl_hda_dsp_generic.HiFi__Speaker__sink Problem summary * The 3.5 mm headphone/line-out jack produces no audio unless the “Headphone” control is manually unmuted via ALSA (alsamixer/amixer). * PipeWire/WirePlumber only expose a single output port (“Speaker”) for this card/profile; there is no separate “Headphones/Line Out” port to select in GNOME/pavucontrol. * Internal speakers do not auto-mute when the external jack is used; some intermittent audio still leaks to the internal speakers. * Jack detection appears unstable (repeated “Headphone Jack” events seen) and the desktop toggles output availability intermittently. Expected outcome * PipeWire/WirePlumber should expose distinct output ports for Speaker and Headphones/Line Out, based on UCM, for this device. * When the 3.5 mm plug is inserted, the system should: 1. Auto-switch to the Headphones/Line Out port, 1. Mute the internal speakers (auto-mute), 1. Keep the state stable (no flapping). 1. Users should not need to manipulate raw ALSA mixer controls manually for basic jack usage. Actual outcome * Only “Speaker” is exposed as an output port; no “Headphones/Line Out” port is available to select. * Audio plays intermittently through internal speakers; external jack is silent until “Headphone” is manually unmuted in alsamixer. * After manual unmute, the external jack works, but internal speakers continue to output intermittently unless manually muted; auto-mute does not engage. * Jack-sense events appear to flap while the plug is steady, causing intermittent switching in the desktop. Steps to reproduce (high level) * Boot system; log in to GNOME (PipeWire/WirePlumber active). * Plug powered external speakers/headphones into the 3.5 mm combo jack. * Observe that "Headphone" output intermittently appears on the dropdown. Hard to select via UI, but when is done, the sound comes out of the internal speakers only, in synchrony with the intermittence in the ui. * Use alsamixer on the SOF HDA card to unmute “Headphone”; external jack starts working, but internal speakers still play unless manually muted; behavior may fluctuate with jack events. Impact: * Users cannot rely on auto-detection or desktop controls for basic headphone/line-out use. Manual ALSA mixer changes are required. Internal speakers may still play, causing privacy/UX issues. Workaround: * Manually unmute “Headphone” and set “Speaker” to lowest volume in alsamixer; Auto-Mute doesn't seem to work in this setup. * Disabling HDA power saving didn't seem to help reduce jack flapping. Possble solution: In alsa-ucm-conf for the SOF HDA “skl_hda_dsp_generic” profile on this Raptor Lake codec: * Define and expose a proper “Headphones/Line Out” device/port alongside “Speaker.” * Add the appropriate Enable/Disable/Jack sequences to unmute Headphone, mute Speaker (auto-mute), and handle jack detection correctly. * If the UCM is already correct for this codec, guidance is appreciated on whether the issue should be addressed in: - ALSA kernel (snd_hda_intel/Realtek codec quirk) or SOF topology/firmware for stable jack sense and correct control exposure. - WirePlumber/PipeWire only if UCM provides the ports but they are not shown. alsa-info: http://alsa-project.org/db/?f=357e5def55a9a2e737306ff2ec22b1526a14f87a Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/632 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

3 2

[GIT PULL] ASoC fixes for v6.18-rc2
by Mark Brown 30 Oct '25

30 Oct '25

The following changes since commit 211ddde0823f1442e4ad052a2f30f050145ccada: Linux 6.18-rc2 (2025-10-19 15:19:16 -1000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.18-rc2 for you to fetch changes up to 5e5c8aa73d99f1daa9f2ec1474b7fc1a6952764b: ASoC: dt-bindings: pm4125-sdw: correct number of soundwire ports (2025-10-29 14:54:47 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.18 A bigger batch of fixes than I'd like, things built up due to holidays and some last minute issues which caused me to hold off on sending a pul request. None of these are super remarkable, and there's a few new device IDs in here too including a relatively big block of AMD devices. The Cirrus Logic CS530x support subject line is actually a fix that was on the start of that series and got pulled in here, I forgot to fix the subject up when merging. ---------------------------------------------------------------- Bard Liao (1): ASoC: soc_sdw_utils: remove cs42l43 component_name Cezary Rojewski (3): ASoC: Intel: avs: Unprepare a stream when XRUN occurs ASoC: Intel: avs: Disable periods-elapsed work when closing PCM ASoC: Intel: avs: Use snd_codec format when initializing probe Claudiu Beznea (1): ASoC: renesas: rz-ssi: Use proper dma_buffer_pos after resume Haotian Zhang (1): ASoC: mediatek: Fix double pm_runtime_disable in remove functions Maarten Zanders (1): ASoC: fsl_sai: Fix sync error in consumer mode Mark Brown (4): Add support for Cirrus Logic CS530x DAC and CODEC ASoC: Intel: avs: Set of streaming fixes ASoC: fsl: correct the bit order issue for DSD ASoC: Fix build for sdw_utils Richard Fitzgerald (1): ASoC: cs-amp-lib-test: Fix missing include of kunit/test-bug.h Sharique Mohammad (1): ASOC: max98090/91: fix for filter configuration: AHPF removed DMIC2_HPF added Shengjiu Wang (2): ASoC: fsl_sai: fix bit order for DSD format ASoC: fsl_micfil: correct the endian format for DSD Shuming Fan (2): ASoC: sdw_utils: add name_prefix for rt1321 part id ASoC: rt721: fix prepare clock stop failed Simon Trimmer (3): ASoC: cs530x: Correct log message with expected variable ASoC: amd: acp: Add ACP7.0 match entries for cs35l56 and cs42l43 ASoC: Intel: soc-acpi-intel-ptl-match: Remove cs42l43 match from sdw link3 Srinivas Kandagatla (2): ASoC: qdsp6: q6asm: do not sleep while atomic ASoC: dt-bindings: pm4125-sdw: correct number of soundwire ports .../devicetree/bindings/sound/qcom,pm4125-sdw.yaml | 4 +- sound/soc/amd/acp/amd-acp70-acpi-match.c | 157 +++++++++++++++++++++ sound/soc/codecs/cs-amp-lib-test.c | 1 + sound/soc/codecs/cs530x.c | 2 +- sound/soc/codecs/max98090.c | 6 +- sound/soc/codecs/rt721-sdca.c | 4 + sound/soc/codecs/rt721-sdca.h | 1 + sound/soc/fsl/fsl_micfil.c | 4 +- sound/soc/fsl/fsl_sai.c | 11 +- sound/soc/intel/avs/pcm.c | 3 + sound/soc/intel/avs/probes.c | 18 +-- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 52 ------- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 1 - sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 1 - sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/renesas/rz-ssi.c | 25 ++-- sound/soc/sdw_utils/soc_sdw_utils.c | 1 - 17 files changed, 203 insertions(+), 90 deletions(-)

2 1

[PATCH AUTOSEL 6.1 02/13] ASoC: amd: yc: Add quirk for Lenovo Yoga Pro 7 14ASP9
by Sasha Levin 30 Oct '25

30 Oct '25

From: Talhah Peerbhai <talhah.peerbhai(a)gmail.com> [ Upstream commit a28206060dc5848a1a2a15b7f6ac6223d869084d ] Similar to many other Lenovo models with AMD chips, the Lenovo Yoga Pro 7 14ASP9 (product name 83HN) requires a specific quirk to ensure internal mic detection. This patch adds a quirk fixing this. Signed-off-by: Talhah Peerbhai <talhah.peerbhai(a)gmail.com> Link: https://patch.msgid.link/20250515222741.144616-1-talhah.peerbhai@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Based on my analysis of this commit, I can provide a comprehensive assessment for backporting. **YES** This commit should be backported to stable kernel trees for the following reasons: **1. Identical Pattern to Previously Backported Commits:** All five similar commits provided in the historical context (commits for Lenovo models 82UU, 82, 82YM, 83AS, and 21M1) have "Backport Status: YES" and follow the exact same pattern as this commit. This establishes a clear precedent for backporting such changes. **2. Hardware Enablement Bug Fix:** The commit fixes a clear user-affecting bug where the internal microphone on the Lenovo Yoga Pro 7 14ASP9 (product name 83HN) is non-functional. This is a hardware compatibility issue that directly impacts users' ability to use their device's built-in microphone. **3. Minimal and Contained Change:** The code change is extremely small and safe: - Adds only a single 7-line quirk table entry to `sound/soc/amd/yc/acp6x-mach.c` - The change is purely additive - no existing code is modified - The quirk entry follows the exact same pattern as dozens of other entries in the same table - Zero risk of regression to existing functionality **4. Critical Subsystem with Safe Pattern:** While this touches audio (a critical subsystem), the change follows a well-established, safe pattern used extensively in this driver. The quirk table approach is designed specifically for adding device-specific compatibility without affecting other hardware. **5. Clear User Benefit:** Users with this specific Lenovo model will have their internal microphone functionality restored, which is essential for video calls, voice recording, and other audio input tasks. **6. Code Quality Improvement:** The commit also includes a minor whitespace fix (changing spaces to tab at line 350), improving code formatting consistency. **7. Follows Stable Tree Rules:** - Important bugfix: ✓ (enables hardware functionality) - Minimal risk: ✓ (purely additive quirk entry) - Small and contained: ✓ (7 lines added) - No architectural changes: ✓ - Confined to subsystem: ✓ (AMD YC audio driver) The commit message clearly explains the issue and solution, and the change is identical in nature to numerous other successfully backported commits for similar Lenovo audio quirks. This represents a textbook example of a stable-appropriate hardware enablement fix. sound/soc/amd/yc/acp6x-mach.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 1f94269e121af..d5dc1d48fca94 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -304,6 +304,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83AS"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83HN"), + } + }, { .driver_data = &acp6x_card, .matches = { @@ -353,7 +360,7 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M5402RA"), } }, - { + { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.39.5

2 1

Qualcomm: qcs615: Add TALOS EVK HiFi config
by GitHub pull_request - edited 29 Oct '25

29 Oct '25

alsa-project/alsa-ucm-conf pull request #631 was edited from mohsRafi: Add UCM2 configs for the Qualcomm TALOS-EVK Board. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/631 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/631.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Qualcomm: qcs616: Add TALOS EVK HiFi config
by GitHub pull_request - edited 29 Oct '25

29 Oct '25

alsa-project/alsa-ucm-conf pull request #631 was edited from mohsRafi: Add UCM2 configs for the Qualcomm TALOS-EVK Board. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/631 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/631.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Qualcomm: qcs616: Add TALOS EVK HiFi config
by GitHub pull_request - opened 29 Oct '25

29 Oct '25

alsa-project/alsa-ucm-conf pull request #631 was opened from mohsRafi: Add UCM2 configs for the Qualcomm Talos Board. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/631 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/631.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Qualcomm: Radxa: fix Displayport SectionDevice
by GitHub pull_request - opened 18 Oct '25

18 Oct '25

alsa-project/alsa-ucm-conf pull request #624 was opened from Srinivas-Kandagatla: for some reason this ucm endedup using DisplayPort for SectionDevice instead of HDMI. This was resulting in below failures of any PR against Qualcomm devices. Run cd alsa-tests/python/ucm-validator2 ./ucm.py configs --level=0 --ucmdir=../../../alsa-ucm-conf/ucm2 ./Qualcomm/qcs6490/QCS6490-Radxa-Dragon-Q6A/HiFi.conf: Device name DisplayPort /'SectionDevice'.'DisplayPort'/ is not valid (see https://github.com/alsa-project/alsa-lib/blob/master/include/use-case.h) total errors: 1 make: *** [Makefile:10: configs] Error 1 Error: Process completed with exit code 2. Fix this by using known devices from use-case.h Fixes: 67628fb4871e ("ucm2: Qualcomm: add Radxa Dragon Q6A") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/624 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/624.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.18-rc1
by Mark Brown 18 Oct '25

18 Oct '25

The following changes since commit 3a8660878839faadb4f1a6dd72c3179c1df56787: Linux 6.18-rc1 (2025-10-12 13:42:36 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.18-rc1 for you to fetch changes up to f1a450f9e17d341f69f8fb19f6d13ef9f1aa508b: ASoC: nau8821: Fix IRQ handling and improve jack (2025-10-16 12:41:35 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.18 A moderately large collection of driver specific fixes, plus a few new quirks and device IDs. The NAU8821 changes are a little large but more in mechanical ways than in ways that are complex. ---------------------------------------------------------------- Baojun Xu (2): ASoC: tas2781: Support more newly-released amplifiers tas58xx in the driver ASoC: tas2781: Update ti,tas2781.yaml for adding tas58xx Christophe Leroy (1): ASoC: codecs: Fix gain setting ranges for Renesas IDT821034 codec Cristian Ciocaltea (5): ASoC: nau8821: Cancel jdet_work before handling jack ejection ASoC: nau8821: Generalize helper to clear IRQ status ASoC: nau8821: Consistently clear interrupts before unmasking ASoC: nau8821: Add DMI quirk to bypass jack debounce circuit ASoC: nau8821: Avoid unnecessary blocking in IRQ handler Frank Li (1): ASoC: dt-bindings: Add compatible string fsl,imx-audio-tlv320 Le Qi (2): ASoC: dt-bindings: qcom,sm8250: Add QCS615 sound card ASoC: qcom: sc8280xp: Add support for QCS615 Li Qiang (1): ASoC: amd/sdw_utils: avoid NULL deref when devm_kasprintf() fails Mark Brown (2): ASoC: Add QCS615 sound card support ASoC: nau8821: Fix IRQ handling and improve jack Sharique Mohammad (1): ASoC: max98090/91: fixed max98091 ALSA widget powering up/down Shuming Fan (1): ASoC: sdw_utils: add rt1321 part id to codec_info_list Srinivas Kandagatla (1): ASoC: codecs: wcd938x-sdw: remove redundant runtime pm calls .../devicetree/bindings/sound/fsl-asoc-card.yaml | 1 + .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + .../devicetree/bindings/sound/ti,tas2781.yaml | 43 ++++++- include/sound/tas2781.h | 3 + sound/soc/amd/acp/acp-sdw-sof-mach.c | 2 +- sound/soc/codecs/idt821034.c | 12 +- sound/soc/codecs/max98090.c | 6 +- sound/soc/codecs/nau8821.c | 129 +++++++++++++-------- sound/soc/codecs/nau8821.h | 2 +- sound/soc/codecs/tas2781-i2c.c | 21 +++- sound/soc/codecs/wcd938x-sdw.c | 20 +--- sound/soc/qcom/sc8280xp.c | 1 + sound/soc/sdw_utils/soc_sdw_utils.c | 20 ++++ 13 files changed, 177 insertions(+), 84 deletions(-)

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Re: [PATCH v4 1/2] ASoC: rt5575: Add the codec driver for the ALC5575
by Mark Brown 18 Oct '25

18 Oct '25

On Mon, Oct 13, 2025 at 05:08:04PM +0800, Oder Chiou wrote: > This codec driver is for the ALC5575 that has a built-in audio DSP. The > firmware can be loaded by the SPI from the SOC or the external flash. > + regmap_update_bits(rt5575->regmap, RT5575_SW_INT, 1, 1); > + ret = 1; > + for (i = 0; i < 100 && ret; i++) { > + regmap_read(rt5575->regmap, RT5575_SW_INT, &ret); > + msleep(100); > + } > + > + return 0; Shouldn't there be an error reported if we get to the end of our 100 loops and ret is still set? > +static int rt5575_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) > +{ > + return 0; > +} Just delete this if it's empty. > +static int rt5575_set_bias_level(struct snd_soc_component *component, > + enum snd_soc_bias_level level) > +{ > + switch (level) { This also does nothing and could be removed. > +static irqreturn_t rt5575_irq(int irq, void *data) > +{ > + return IRQ_HANDLED; > +} This seems strange? Why even register an interrupt handler if we're just going to ignore it?

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Re: [PATCH v4 1/2] ASoC: tas2781: Support more newly-released amplifiers tas58xx in the driver
by Mark Brown 18 Oct '25

18 Oct '25

On Thu, 02 Oct 2025 15:29:24 +0800, Baojun Xu wrote: > TAS5802/TAS5815/TAS5828 has on-chip DSP without current/voltage feedback. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/2] ASoC: tas2781: Support more newly-released amplifiers tas58xx in the driver commit: 53a3c6e222836a23e8e0693395584aefc456dca6 [2/2] ASoC: tas2781: Update ti,tas2781.yaml for adding tas58xx commit: 7e6cfa3e94cf8278ee3170dc0b81fc6db1287e28 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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ucm2: Add support for MT8196 Rauru Rev0 Chromebook with SOF
by GitHub pull_request - opened 18 Oct '25

18 Oct '25

alsa-project/alsa-ucm-conf pull request #625 was opened from laura-nao: Add support for the Google Rauru Rev0 Chromebook, powered by MediaTek Kompanio Ultra (MT8196). This machine uses NAU8825 as headphone codec and NAU8318 as speaker codec. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/625 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/625.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Missing UCM matcher for ASUS Zenbook S 14 UX5406SA (cs42l43/cs35l56, Lunar Lake)
by GitHub issues - edited 26 Sep '25

26 Sep '25

alsa-project/alsa-ucm-conf issue #619 was edited from sjolo312: This is not a kernel bug but a missing UCM matcher. Result: With the matcher below and the latest upstream alsa-ucm-conf files, speakers and microphone work on the ASUS Zenbook S 14 UX5406SA (Lunar Lake, cs42l43 + cs35l56). Tested on: - Ubuntu 25.10 beta (up-to-date live session) - alsa-ucm-conf: upstream master from GitHub (as of 2025-09-26) CardLongName: `ASUSTeKCOMPUTERINC.-ASUSZenbookS14UX5406SA_UX5406SA-1.0-UX5406SA` Root cause: Upstream master does not contain a matcher for this CardLongName, so no UCM profile is loaded (-> “Dummy Output”). Adding the matcher below ensures the existing sof-soundwire profile (which already includes cs42l43/cs35l56 routes) is loaded, and audio works. Proposed new file: Path: `ucm2/conf.d/sof-soundwire/ASUS-Zenbook-S14-UX5406SA.conf` ``` # ASUS Zenbook S 14 UX5406SA (Lunar Lake) SectionDefaults."sof-soundwire" { Condition { Type Regex CardLongName "ASUSTeKCOMPUTERINC.-ASUSZenbookS14UX5406SA.*" } Include.conf "sof-soundwire/sof-soundwire.conf" } ``` Verification (files actually loaded): Gathered with: `strace -f -e trace=%file -o /tmp/ucm.trace \ alsaucm -c sof-soundwire list _verbs >/dev/null 2>&1 grep -oE '/usr/share/alsa/ucm2/[^"]+' /tmp/ucm.trace | sort -u ` Resulting files included: ` /usr/share/alsa/ucm2/Intel/sof-hda-dsp/dsp.conf /usr/share/alsa/ucm2/blobs/sof/product_configs/ASUSTeK COMPUTER INC./ASUS Zenbook S 14 UX5406SA_UX5406SA.conf /usr/share/alsa/ucm2/blobs/sof/user_configs/ASUSTeK COMPUTER INC./ASUS Zenbook S 14 UX5406SA_UX5406SA.conf /usr/share/alsa/ucm2/codecs/cs35l56/init.conf /usr/share/alsa/ucm2/codecs/cs42l43/init.conf /usr/share/alsa/ucm2/codecs/hda/hdmi.conf /usr/share/alsa/ucm2/common/ctl/led.conf /usr/share/alsa/ucm2/common/pcm/hdmi.conf /usr/share/alsa/ucm2/conf.d/sof-soundwire/ASUSTeKCOMPUTERINC.-ASUSZenbookS14UX5406SA_UX5406SA-1.0-UX5406SA.conf /usr/share/alsa/ucm2/conf.d/sof-soundwire/sof-soundwire.conf /usr/share/alsa/ucm2/lib/card-init.conf /usr/share/alsa/ucm2/lib/ctl-remap.conf /usr/share/alsa/ucm2/lib/generic.conf /usr/share/alsa/ucm2/sof-soundwire/Hdmi.conf /usr/share/alsa/ucm2/sof-soundwire/HiFi.conf /usr/share/alsa/ucm2/sof-soundwire/cs35l56.conf /usr/share/alsa/ucm2/sof-soundwire/cs42l43.conf /usr/share/alsa/ucm2/sof-soundwire/dmic.conf /usr/share/alsa/ucm2/ucm.conf ` Request: Please add this matcher to alsa-ucm-conf upstream, so that the ASUS Zenbook S 14 UX5406SA works out-of-the-box on all Linux distributions. Happy to test any proposed patch or provide further logs if needed. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/619 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Missing UCM matcher for ASUS Zenbook S 14 UX5406SA (cs42l43/cs35l56, Lunar Lake)
by GitHub issues - opened 26 Sep '25

26 Sep '25

alsa-project/alsa-ucm-conf issue #619 was opened from sjolo312: This is not a kernel bug but a missing UCM matcher. Result: With the matcher below and the latest upstream alsa-ucm-conf files, speakers and microphone work on the ASUS Zenbook S 14 UX5406SA (Lunar Lake, cs42l43 + cs35l56). Tested on: - Ubuntu 25.10 beta (up-to-date live session) - alsa-ucm-conf: upstream master from GitHub (as of 2025-09-26) CardLongName: ASUSTeKCOMPUTERINC.-ASUSZenbookS14UX5406SA_UX5406SA-1.0-UX5406SA Root cause: Upstream master does not contain a matcher for this CardLongName, so no UCM profile is loaded (-> “Dummy Output”). Adding the matcher below ensures the existing sof-soundwire profile (which already includes cs42l43/cs35l56 routes) is loaded, and audio works. Proposed new file: Path: ucm2/conf.d/sof-soundwire/ASUS-Zenbook-S14-UX5406SA.conf # ASUS Zenbook S 14 UX5406SA (Lunar Lake) SectionDefaults."sof-soundwire" { Condition { Type Regex CardLongName "ASUSTeKCOMPUTERINC.-ASUSZenbookS14UX5406SA.*" } Include.conf "sof-soundwire/sof-soundwire.conf" } Verification (files actually loaded): Gathered with: strace -f -e trace=%file -o /tmp/ucm.trace \ alsaucm -c sof-soundwire list _verbs >/dev/null 2>&1 grep -oE '/usr/share/alsa/ucm2/[^"]+' /tmp/ucm.trace | sort -u Resulting files included: /usr/share/alsa/ucm2/Intel/sof-hda-dsp/dsp.conf /usr/share/alsa/ucm2/blobs/sof/product_configs/ASUSTeK COMPUTER INC./ASUS Zenbook S 14 UX5406SA_UX5406SA.conf /usr/share/alsa/ucm2/blobs/sof/user_configs/ASUSTeK COMPUTER INC./ASUS Zenbook S 14 UX5406SA_UX5406SA.conf /usr/share/alsa/ucm2/codecs/cs35l56/init.conf /usr/share/alsa/ucm2/codecs/cs42l43/init.conf /usr/share/alsa/ucm2/codecs/hda/hdmi.conf /usr/share/alsa/ucm2/common/ctl/led.conf /usr/share/alsa/ucm2/common/pcm/hdmi.conf /usr/share/alsa/ucm2/conf.d/sof-soundwire/ASUSTeKCOMPUTERINC.-ASUSZenbookS14UX5406SA_UX5406SA-1.0-UX5406SA.conf /usr/share/alsa/ucm2/conf.d/sof-soundwire/sof-soundwire.conf /usr/share/alsa/ucm2/lib/card-init.conf /usr/share/alsa/ucm2/lib/ctl-remap.conf /usr/share/alsa/ucm2/lib/generic.conf /usr/share/alsa/ucm2/sof-soundwire/Hdmi.conf /usr/share/alsa/ucm2/sof-soundwire/HiFi.conf /usr/share/alsa/ucm2/sof-soundwire/cs35l56.conf /usr/share/alsa/ucm2/sof-soundwire/cs42l43.conf /usr/share/alsa/ucm2/sof-soundwire/dmic.conf /usr/share/alsa/ucm2/ucm.conf Request: Please add this matcher to alsa-ucm-conf upstream, so that the ASUS Zenbook S 14 UX5406SA works out-of-the-box on all Linux distributions. Happy to test any proposed patch or provide further logs if needed. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/619 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2 1/2] ASoC: tas2781: Add tas5828 support
by Baojun Xu 26 Sep '25

26 Sep '25

TAS5828 have on-chip DSP without current/voltage feedback. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2: - Update description for TAS5828 - Change commit tree to .../tiwai/sound.git. --- include/sound/tas2781.h | 1 + sound/soc/codecs/tas2781-i2c.c | 7 ++++++- 2 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index ddd997ac3..a37b4d67c 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -122,6 +122,7 @@ enum audio_device { TAS2781, TAS5825, TAS5827, + TAS5828, TAS_OTHERS, }; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 1539b7088..f62a71dca 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -110,6 +110,7 @@ static const struct i2c_device_id tasdevice_id[] = { { "tas2781", TAS2781 }, { "tas5825", TAS5825 }, { "tas5827", TAS5827 }, + { "tas5828", TAS5828 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); @@ -126,6 +127,7 @@ static const struct of_device_id tasdevice_of_match[] = { { .compatible = "ti,tas2781" }, { .compatible = "ti,tas5825" }, { .compatible = "ti,tas5827" }, + { .compatible = "ti,tas5828" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -1665,7 +1667,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, } tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - /* There is no calibration required for TAS5825/TAS5827. */ + /* There is no calibration required for TAS5825/TAS5827/TAS5828. */ if (tas_priv->chip_id < TAS5825) { ret = tasdevice_create_cali_ctrls(tas_priv); if (ret) { @@ -1722,6 +1724,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, case TAS2781: case TAS5825: case TAS5827: + case TAS5828: /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } @@ -1884,6 +1887,7 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) break; case TAS5825: case TAS5827: + case TAS5828: p = (struct snd_kcontrol_new *)tas5825_snd_controls; size = ARRAY_SIZE(tas5825_snd_controls); break; @@ -2056,6 +2060,7 @@ static const struct acpi_device_id tasdevice_acpi_match[] = { { "TXNW2781", TAS2781 }, { "TXNW5825", TAS5825 }, { "TXNW5827", TAS5827 }, + { "TXNW5828", TAS5828 }, {}, }; -- 2.25.1

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ucm2: Qualcomm: sa8775p: Move lemans-evk HiFi to sa8775p subdir
by GitHub pull_request - opened 26 Sep '25

26 Sep '25

alsa-project/alsa-ucm-conf pull request #618 was opened from mohsRafi: Move lemans-evk HiFi configuration files to the sa8775p subdirectory, aligning with the kernel's use of the sa8775p folder for all Linux firmware. So Update the UCM HiFi configuration file accordingly. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/618 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/618.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Looking for inputs on hostless playback with the DSP
by Sridharan, Ranjani 26 Sep '25

26 Sep '25

Hi Mark/Takashi, I would like to implement a topology with a tone generator widget running in the DSP that is connected to the speaker codec at the backend. So, my pipeline looks something like this: Tone Generator Module (DSP) -> DAI Component (DSP) -> Speaker Codec Without a PCM frontend, how would I go about activating this path in my topology? While I can think of adding a kcontrol for the tone generator widget to enable the pipeline in the DSP, I'm wondering how to do the hw_params, startup/shutdown for the backend speaker codec. Could you please help me with some pointers on how to do this? Thanks, Ranjani

2 3

ASOC - Manage sound card with multiple optional codecs
by claudio.mocci＠abinsula.com 26 Sep '25

26 Sep '25

I'm working on a custom driver for an embedded device that should manage an audio card with multiple codecs. Each codec can be hotplugged or removed because is placed on an external display that extends an i2cbus. I used the orginal fsl/imx-card.c driver as a starting point and my device tree looks like this: sound_card: sound { compatible = "display-audio-card"; model = "disp-audio"; pri-dai-link { link-name = "disp-codec-1"; format = "dsp_b"; dai-tdm-slot-num = <4>; dai-tdm-slot-width = <32>; dai-tdm-slot-tx-mask = <1 1 1 1 0 0 0 0>; fsl,mclk-equal-bclk; cpu { sound-dai = <&esai0>; }; codec { sound-dai = <&speaker_dac_1>, <&speaker_dac_2>; }; }; }; The first issue I encountered was that if both of the codecs were not present at the system start the snd_soc_of_get_dai_link_codecs was returning probe_defer, so I made an internal customization to set one of the two codecs to dummy and continuing registering the card that works as required. Now the problem I am facing is when I have both displays connected and when I unplug one of them all the card gets unregistered instead of the single codec. I tried implementing the codec_i2c_remove where I removed the component from the component list snd_soc_component_remove(component); list_del(&component->list); devm_kfree(dev, component->name); and this did not unregistered the card, but something is odd. So I tried to replace the component with a dummy codec to be transparent but without success There is some other architecture instead of imx-card that I can use to implement the feature? I remember firsly I tried with audio-graph-card but without success.

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Re: [PATCH v2 2/2] ASoC: tas2781: Add tas5828 support
by Mark Brown 26 Sep '25

26 Sep '25

On Fri, Sep 26, 2025 at 12:33:39PM +0800, Baojun Xu wrote: > Update ti,tas2781.yaml for adding TAS5828. Reviewed-by: Mark Brown <broonie(a)kernel.org>

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[PATCH v1 1/2] ASoC: tas2781: Add tas5828 support
by Baojun Xu 25 Sep '25

25 Sep '25

TAS5828 have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781.h | 1 + sound/soc/codecs/tas2781-i2c.c | 7 ++++++- 2 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index ddd997ac3216..a37b4d67c137 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -122,6 +122,7 @@ enum audio_device { TAS2781, TAS5825, TAS5827, + TAS5828, TAS_OTHERS, }; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 1539b70881d1..f62a71dca36c 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -110,6 +110,7 @@ static const struct i2c_device_id tasdevice_id[] = { { "tas2781", TAS2781 }, { "tas5825", TAS5825 }, { "tas5827", TAS5827 }, + { "tas5828", TAS5828 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); @@ -126,6 +127,7 @@ static const struct of_device_id tasdevice_of_match[] = { { .compatible = "ti,tas2781" }, { .compatible = "ti,tas5825" }, { .compatible = "ti,tas5827" }, + { .compatible = "ti,tas5828" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -1665,7 +1667,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, } tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - /* There is no calibration required for TAS5825/TAS5827. */ + /* There is no calibration required for TAS5825/TAS5827/TAS5828. */ if (tas_priv->chip_id < TAS5825) { ret = tasdevice_create_cali_ctrls(tas_priv); if (ret) { @@ -1722,6 +1724,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, case TAS2781: case TAS5825: case TAS5827: + case TAS5828: /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } @@ -1884,6 +1887,7 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) break; case TAS5825: case TAS5827: + case TAS5828: p = (struct snd_kcontrol_new *)tas5825_snd_controls; size = ARRAY_SIZE(tas5825_snd_controls); break; @@ -2056,6 +2060,7 @@ static const struct acpi_device_id tasdevice_acpi_match[] = { { "TXNW2781", TAS2781 }, { "TXNW5825", TAS5825 }, { "TXNW5827", TAS5827 }, + { "TXNW5828", TAS5828 }, {}, }; -- 2.43.0

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[PATCH v1] slimbus: qcom-ngd-ctrl: Add stream disable support
by Viken Dadhaniya 24 Sep '25

24 Sep '25

Currently slimbus driver doesn't support stream disable callback, it only supports stream enable callback. In slimbus usecase, client is switching to new frequency with same channel and calling enable stream callback for new frequency but DSP subsystem is crashing as we are switching to new frequency with same channel without disabling stream for older frequency. Ideally, before switching to another frequency, client should call disable stream callback and then enable stream for newer frequency. Hence add support to disable stream via qcom_slim_ngd_disable_stream(). Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 70 +++++++++++++++++++++++++++++++++ drivers/slimbus/slimbus.h | 13 ++++++ 2 files changed, 83 insertions(+) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index e0b21f0f79c1..d952827d2e12 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1,6 +1,7 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited +// Copyright (c) 2024 Qualcomm Innovation Center, Inc. All rights reserved. #include <linux/irq.h> #include <linux/kernel.h> @@ -1084,6 +1085,74 @@ static int qcom_slim_ngd_enable_stream(struct slim_stream_runtime *rt) return ret; } +static int qcom_slim_ngd_disable_stream(struct slim_stream_runtime *rt) +{ + struct slim_device *sdev = rt->dev; + struct slim_controller *ctrl = sdev->ctrl; + struct slim_val_inf msg = {0}; + u8 wbuf[SLIM_MSGQ_BUF_LEN]; + u8 rbuf[SLIM_MSGQ_BUF_LEN]; + struct slim_msg_txn txn = {0,}; + int i, ret; + + txn.mt = SLIM_MSG_MT_DEST_REFERRED_USER; + txn.dt = SLIM_MSG_DEST_LOGICALADDR; + txn.la = SLIM_LA_MGR; + txn.ec = 0; + txn.msg = &msg; + txn.msg->num_bytes = 0; + txn.msg->wbuf = wbuf; + txn.msg->rbuf = rbuf; + + for (i = 0; i < rt->num_ports; i++) { + struct slim_port *port = &rt->ports[i]; + + if (txn.msg->num_bytes == 0) { + wbuf[txn.msg->num_bytes++] = (u8)(SLIM_CH_REMOVE << 6) + | (sdev->laddr & 0x1f); + + ret = slim_alloc_txn_tid(ctrl, &txn); + if (ret) { + dev_err(&sdev->dev, "Fail to allocate TID\n"); + return -ENXIO; + } + wbuf[txn.msg->num_bytes++] = txn.tid; + } + wbuf[txn.msg->num_bytes++] = port->ch.id; + } + + txn.mc = SLIM_USR_MC_CHAN_CTRL; + txn.rl = txn.msg->num_bytes + 4; + ret = qcom_slim_ngd_xfer_msg_sync(ctrl, &txn); + if (ret) { + slim_free_txn_tid(ctrl, &txn); + dev_err(&sdev->dev, "TX timed out:MC:0x%x,mt:0x%x ret:%d\n", + txn.mc, txn.mt, ret); + return ret; + } + + txn.mc = SLIM_USR_MC_RECONFIG_NOW; + txn.msg->num_bytes = 2; + wbuf[1] = sdev->laddr; + txn.rl = txn.msg->num_bytes + 4; + + ret = slim_alloc_txn_tid(ctrl, &txn); + if (ret) { + dev_err(ctrl->dev, "Fail to allocate TID ret:%d\n", ret); + return ret; + } + + wbuf[0] = txn.tid; + ret = qcom_slim_ngd_xfer_msg_sync(ctrl, &txn); + if (ret) { + slim_free_txn_tid(ctrl, &txn); + dev_err(&sdev->dev, "TX timed out:MC:0x%x,mt:0x%x ret:%d\n", + txn.mc, txn.mt, ret); + } + + return ret; +} + static int qcom_slim_ngd_get_laddr(struct slim_controller *ctrl, struct slim_eaddr *ea, u8 *laddr) { @@ -1642,6 +1711,7 @@ static int qcom_slim_ngd_ctrl_probe(struct platform_device *pdev) ctrl->ctrl.clkgear = SLIM_MAX_CLK_GEAR; ctrl->ctrl.get_laddr = qcom_slim_ngd_get_laddr; ctrl->ctrl.enable_stream = qcom_slim_ngd_enable_stream; + ctrl->ctrl.disable_stream = qcom_slim_ngd_disable_stream; ctrl->ctrl.xfer_msg = qcom_slim_ngd_xfer_msg; ctrl->ctrl.wakeup = NULL; ctrl->state = QCOM_SLIM_NGD_CTRL_DOWN; diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 00a7f112574b..21543d125614 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -1,6 +1,7 @@ /* SPDX-License-Identifier: GPL-2.0 */ /* * Copyright (c) 2011-2017, The Linux Foundation + * Copyright (c) 2024 Qualcomm Innovation Center, Inc. All rights reserved. */ #ifndef _DRIVERS_SLIMBUS_H @@ -316,6 +317,18 @@ enum slim_transport_protocol { SLIM_PROTO_EXT_HALF_DUP, }; +/* + * enum slim_ch_control: Channel control. + * Activate will schedule channel and/or group of channels in the TDM frame. + * Suspend will keep the schedule but data-transfer won't happen. + * Remove will remove the channel/group from the TDM frame. + */ +enum slim_ch_control { + SLIM_CH_ACTIVATE, + SLIM_CH_SUSPEND, + SLIM_CH_REMOVE, +}; + /** * struct slim_stream_runtime - SLIMbus stream runtime instance * -- 2.17.1

2 2

UCM Error for MOTU M6
by GitHub issues - opened 23 Sep '25

23 Sep '25

alsa-project/alsa-ucm-conf issue #617 was opened from FreddieWitherden: My USB MOTU M6 USB device works fine, but the KDE sound settings dialogue reports it as "M Series [ALSA UCM error] Pro" in both the Playback Devices and Recording Devices. I've attached the output of `alsa-info.sh --no-upload`. [alsa-info.txt](https://github.com/user-attachments/files/22492752/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/617 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ASoC: tas2781: Correct the wrong description and register address on tas2781
by Baojun Xu 22 Sep '25

22 Sep '25

Correct the wrong description for TAS257X. Combined TAS5825 with TAS2563, as they use the same register address and number. Correct the register address and number for TAS5827. Fixes: 7095d688de38 ("ASoC: tas2781: Add tas2118, tas2x20, tas5825 support") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- .../devicetree/bindings/sound/ti,tas2781.yaml | 20 +++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml index 011211112be4..bd00afa47d62 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml +++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml @@ -11,11 +11,13 @@ maintainers: - Shenghao Ding <shenghao-ding(a)ti.com> description: | - The TAS2118/TAS2X20/TAS257x is mono, digital input Class-D audio + The TAS2118/TAS2X20 is mono, digital input Class-D audio amplifier optimized for efficiently driving high peak power into small loudspeakers. - Integrated speaker voltage and current sense provides for - real time monitoring of loudspeaker behavior. + The TAS257x is mono, digital input Class-D audio amplifier optimized + for efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for real time + monitoring of loudspeaker behavior. The TAS2563/TAS2781 is a mono, digital input Class-D audio amplifier optimized for efficiently driving high peak power into small loudspeakers. An integrated on-chip DSP supports Texas @@ -25,9 +27,7 @@ description: | The TAS5825/TAS5827 is a stereo, digital input Class-D audio amplifier optimized for efficiently driving high peak power into small loudspeakers. An integrated on-chip DSP supports Texas - Instruments Smart Amp speaker protection algorithm. The - integrated speaker voltage and current sense provides for real time - monitoring of loudspeaker behavior. + Instruments Smart Amp speaker protection algorithm. Specifications about the audio amplifier can be found at: https://www.ti.com/lit/gpn/tas2120 @@ -131,6 +131,7 @@ allOf: contains: enum: - ti,tas2563 + - ti,tas5825 then: properties: reg: @@ -181,15 +182,14 @@ allOf: compatible: contains: enum: - - ti,tas5825 - ti,tas5827 then: properties: reg: - maxItems: 4 + maxItems: 6 items: - minimum: 0x4c - maximum: 0x4f + minimum: 0x60 + maximum: 0x65 additionalProperties: false -- 2.43.0

1 0

Debian 13, MacBookPro 13, T2, 6.16.6. Steroe Line SSL2 Alsa UCM error but working fine
by GitHub issues - opened 21 Sep '25

21 Sep '25

alsa-project/alsa-ucm-conf issue #616 was opened from brandtan: Hi all, I am using debian 13 on my macbook pro 15,2 T2. Audio subsystem is working fine. However, there is the issue appearing: UCM error These are my alsa information: http://alsa-project.org/db/?f=e4cf3e09c303e10315cb2a970c85af98ccb80a53 Hope that there will be a bugfix soon. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/616 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS Proart P16 - Realtek ALC245 not detected properly after kernel 6.15 - worked perfectly with earlier kernels
by GitHub issues - edited 21 Sep '25

21 Sep '25

alsa-project/alsa-ucm-conf issue #615 was edited from SneakyLittleman: Hey there, as [reported in the arch forums](https://bbs.archlinux.org/viewtopic.php?id=307540), the ALC245 (and Cirrus Logic amp CS35L56) are not detected properly after kernel 6.15. Anything below 6.14.9 works perfectly fine. The chip reverts to the "proaudio" profile, with no other option, and the command "alsaucm listcards" returns an empty list. With older kernels, the command returned: ` 0: hw:1 HDA NVidia at 0xdd080000 irq 143 1: hw:2 HD-Audio Generic at 0xdd7c8000 irq 144 2: hw:3 HD-Audio Generic at 0xdd7c0000 irq 145 3: hw:4 ASUSTeKCOMPUTERINC.-ProArtP16H7606WI_H7606WI-1.0-H7606WI` and I had access to the correct audio profiles with pipewire. Willing to submit anything that might help, to fix this. Cheers. Here's the chip info in that specific laptop: `66:00.6 Audio device [0403]: Advanced Micro Devices, Inc. [AMD] Family 17h/19h/1ah HD Audio Controller [1022:15e3] DeviceName: Realtek ALC245 Subsystem: ASUSTeK Computer Inc. Device [1043:1e63]` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/615 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS Proart P16 - Realtek ALC245 not detected properly after kernel 6.15 - worked perfectly with earlier kernels
by GitHub issues - edited 21 Sep '25

21 Sep '25

alsa-project/alsa-ucm-conf issue #615 was edited from SneakyLittleman: Hey there, as [reported in the arch forums](https://bbs.archlinux.org/viewtopic.php?id=307540), the ALC285 (and Cirrus Logic amp CS35L56) are not detected properly after kernel 6.15. Anything below 6.14.9 works perfectly fine. The chip reverts to the "proaudio" profile, with no other option, and the command "alsaucm listcards" returns an empty list. With older kernels, the command returned: ` 0: hw:1 HDA NVidia at 0xdd080000 irq 143 1: hw:2 HD-Audio Generic at 0xdd7c8000 irq 144 2: hw:3 HD-Audio Generic at 0xdd7c0000 irq 145 3: hw:4 ASUSTeKCOMPUTERINC.-ProArtP16H7606WI_H7606WI-1.0-H7606WI` and I had access to the correct audio profiles with pipewire. Willing to submit anything that might help, to fix this. Cheers. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/615 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS Proart P16 - Realtek ALC285 not detected properly after kernel 6.15 - worked perfectly with earlier kernels
by GitHub issues - opened 21 Sep '25

21 Sep '25

alsa-project/alsa-ucm-conf issue #615 was opened from SneakyLittleman: Hey there, as [reported in the arch forums](https://bbs.archlinux.org/viewtopic.php?id=307540), the ALC285 (and Cirrus Logic amp CS35L56) are not detected properly after kernel 6.15. Anything below 6.14.9 works perfectly fine. The chip reverts to the "proaudio" profile, with no other option, and the command "alsaucm listcards" returns an empty list. With older kernels, the command returned: ` 0: hw:1 HDA NVidia at 0xdd080000 irq 143 1: hw:2 HD-Audio Generic at 0xdd7c8000 irq 144 2: hw:3 HD-Audio Generic at 0xdd7c0000 irq 145 3: hw:4 ASUSTeKCOMPUTERINC.-ProArtP16H7606WI_H7606WI-1.0-H7606WI` and I had access to the correct audio profiles with pipewire. Willing to submit anything that might help, to fix this. Cheers. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/615 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo 83KF with Realtek ALC287 – bass speaker not exposed (only treble on Linux, Dolby works on Windows)
by GitHub issues - edited 20 Sep '25

20 Sep '25

alsa-project/alsa-ucm-conf issue #614 was edited from Prajwal-Prathiksh: > [!IMPORTANT] > **I just started getting into Linux, and I'm not even sure if this is the right place to be posting this issue. I'm more than willing to learn or improve if I need to be doing something else or reaching out elsewhere :")** ### System - Manufacturer: LENOVO - Product Name: 83KF - Product Version: Yoga Pro 7 14IAH10 - Firmware Version: QGCN28WW - System SKU: LENOVO_MT_83KF_BU_idea_FM_Yoga Pro 7 14IAH10 - Board Vendor: LENOVO - Board Name: LNVNB161216 - Kernel: 6.16.8-arch1-1 - Packages: - sof-firmware 2025.05.1-1 - alsa-ucm-conf 1.2.14-2 - alsa-utils 1.2.14-1 - Audio stack: PipeWire (Arch Linux) ### Problem - On Windows with Dolby Atmos, laptop speakers produce rich sound with bass. - On Linux (Arch), sound is thin/treble-only. No bass speaker output. - In `alsamixer`, only `Speaker` control is visible. - Codec dump shows: - 0x14: [Fixed] Internal Speaker (plays sound, treble only) - 0x17/0x1b/0x1e: [N/A] Speaker at Ext Rear (do not output audio) ### What I tried - Updated `sof-firmware` + `alsa-ucm-conf`. - Raised mixer levels. - Tested with `hda-verb` to enable 0x17/0x1b as output, set gain, EAPD, and connect to DAC → no sound. - Confirmed with `speaker-test -t sine -f 80` and audio tracks from YouTube. ### Logs Attaching files - [alsa-info.txt](https://github.com/user-attachments/files/22440886/alsa-info… - [amixer_contents.txt](https://github.com/user-attachments/files/22440883/ami… - [amixer_simple.txt](https://github.com/user-attachments/files/22440885/amixe… - [codec_dump.txt](https://github.com/user-attachments/files/22440884/codec_du… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/614 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo 83KF with Realtek ALC287 – bass speaker not exposed (only treble on Linux, Dolby works on Windows)
by GitHub issues - opened 20 Sep '25

20 Sep '25

alsa-project/alsa-ucm-conf issue #614 was opened from Prajwal-Prathiksh: > [!IMPORTANT] > **I just started getting into Linux, and I'm not even sure if this is the right place to be posting this issue. I'm more than willing to learn or improve if I need to be doing something else or reaching out elsewhere :")** ### System - Manufacturer: LENOVO - Product Name: 83KF - Product Version: Yoga Pro 7 14IAH10 - Firmware Version: QGCN28WW - System SKU: LENOVO_MT_83KF_BU_idea_FM_Yoga Pro 7 14IAH10 - Board Vendor: LENOVO - Board Name: LNVNB161216 - Kernel: 6.16.8-arch1-1 - Packages: - sof-firmware 2025.05.1-1 - alsa-ucm-conf 1.2.14-2 - alsa-utils 1.2.14-1 - Audio stack: PipeWire (Arch Linux) ### Problem - On Windows with Dolby Atmos, laptop speakers produce rich sound with bass. - On Linux (Arch), sound is thin/treble-only. No bass speaker output. - In `alsamixer`, only `Speaker` control is visible. - Codec dump shows: - 0x14: [Fixed] Internal Speaker (plays sound, treble only) - 0x17/0x1b/0x1e: [N/A] Speaker at Ext Rear (do not output audio) ### What I tried - Updated `sof-firmware` + `alsa-ucm-conf`. - Raised mixer levels. - Tested with `hda-verb` to enable 0x17/0x1b as output, set gain, EAPD, and connect to DAC → no sound. - Confirmed with `speaker-test -t sine -f 80`. ### Logs Attaching files - [alsa-info.txt](https://github.com/user-attachments/files/22440886/alsa-info… - [amixer_contents.txt](https://github.com/user-attachments/files/22440883/ami… - [amixer_simple.txt](https://github.com/user-attachments/files/22440885/amixe… - [codec_dump.txt](https://github.com/user-attachments/files/22440884/codec_du… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/614 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC updates for v6.18
by Mark Brown 20 Sep '25

20 Sep '25

The following changes since commit f83ec76bf285bea5727f478a68b894f5543ca76e: Linux 6.17-rc6 (2025-09-14 14:21:14 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.18 for you to fetch changes up to 5998f0d07d2c4bf3d8ecc4e6218e6a324aeb7301: ASoC: fsl: fsl_qmc_audio: Reduce amount of (2025-09-19 21:23:48 +0100) ---------------------------------------------------------------- ASoC: Updates for v6.18 A relatively quiet release for ASoC, we've had a lot of maintainance work going on and several new drivers but really the most remarkable thing is that we removed a driver, the WL1273 driver used in some old Nokia systems that have had the underlying system support removed from the kernel. - Morimoto-san continues his work on cleanups of the core APIs and enforcement of abstraction layers. - Lots of cleanups and conversions of DT bindings. - Substantial maintainance work on the Intel AVS drivers. - Support for Qualcomm Glymur and PM4125, Realtek RT1321, Shanghai FourSemi FS2104/5S, Texas Instruments PCM1754. - Remove support for TI WL1273. ---------------------------------------------------------------- Alex Tran (3): ASoC: codecs: tlv320dac33: Remove unused struct tlv320dac33_platform_data and header file tlv320dac33-plat.h ASoC: codecs: tlv320dac33: Add default value for burst_bclkdiv ASoC: codecs: tlv320dac33: Convert to use gpiod api Alexey Klimov (4): ASoC: codecs: add new pm4125 audio codec driver MAINTAINERS: add Qualcomm PM4125 audio codec to drivers list ASoC: dt-bindings: add bindings for pm4125 audio codec ASoC: qcom: sc8280xp: explicitly set S16LE format in sc8280xp_be_hw_params_fixup() Alvin Šipraga (1): ASoC: codecs: pcm1754: add pcm1754 dac driver Andy Shevchenko (1): ASoC: amd: acp: Remove (explicitly) unused header Arnd Bergmann (2): ASoC: add GPIOLIB_LEGACY dependency where needed ASoC: pxa: add GPIOLIB_LEGACY dependency Bagas Sanjaya (1): ASoC: doc: Internally link to Writing an ALSA Driver docs Bard Liao (3): ASoC: SOF: Intel: split codec mask detect and programming UNSOL ASoC: SOF: Intel: only detect codecs when HDA DSP probe ASoC: Intel: hda-sdw-bpt: set persistent_buffer false Cezary Rojewski (28): ASoC: Intel: avs: Parse conditional path tuples ASoC: Intel: avs: Conditional-path support ASoC: Intel: avs: Separate debug symbols ASoC: Intel: avs: Drop pcm.h dependency for probes ASoC: Intel: avs: Do not share the name pointer between components ASoC: Intel: avs: Streamline register-component function names ASoC: Intel: avs: New board registration routines ASoC: Intel: avs: Cleanup duplicate members ASoC: Intel: avs: Simplify dmi_match_quirk() ASoC: Intel: avs: Simplify avs_get_i2s_boards() ASoC: Intel: avs: Allow i2s test and non-test boards to coexist ASoC: Intel: avs: hda: Adjust platform name ASoC: Intel: avs: da7219: Adjust platform name ASoC: Intel: avs: dmic: Refactor dai_link creation ASoC: Intel: avs: es8336: Adjust platform name ASoC: Intel: avs: i2stest: Adjust platform name ASoC: Intel: avs: max98357a: Adjust platform name ASoC: Intel: avs: max98373: Adjust platform name ASoC: Intel: avs: max98927: Adjust platform name ASoC: Intel: avs: nau8825: Adjust platform name ASoC: Intel: avs: probe: Refactor dai_link creation ASoC: Intel: avs: rt274: Adjust platform name ASoC: Intel: avs: rt286: Adjust platform name ASoC: Intel: avs: rt298: Adjust platform name ASoC: Intel: avs: rt5514: Adjust platform name ASoC: Intel: avs: rt5663: Adjust platform name ASoC: Intel: avs: rt5682: Adjust platform name ASoC: Intel: avs: ssm4567: Adjust platform name Chancel Liu (1): ASoC: dmaengine_pcm: Add port_window_size to DAI dma data struct Charles Han (1): ASoC: codecs: fs210x: Add NULL check in fs210x_register_snd_component Charles Keepax (4): ASoC: cs42l43: Rename system suspend callback and fix debug print ASoC: cs42l43: Store IRQ domain in codec private data ASoC: cs42l43: Disable IRQs in system suspend ASoC: cs42l43: Shutdown jack detection on suspend Chen-Yu Tsai (1): ASoC: mediatek: common: Switch to for_each_available_child_of_node_scoped() Christophe Leroy (4): soc: fsl: qmc: Only set completion interrupt when needed ASoC: fsl: fsl_qmc_audio: Ensure audio channels are ordered in TDM bus ASoC: fsl: fsl_qmc_audio: Only request completion on last channel ASoC: fsl: fsl_qmc_audio: Drop struct qmc_dai_chan Colin Ian King (1): ASoC: SOF: ipc4-topology: Fix a less than zero check on a u32 Dr. David Alan Gilbert (1): ASoC: wl1273: Remove Geert Uytterhoeven (1): ASoC: da7213: Convert to DEFINE_RUNTIME_DEV_PM_OPS() Jihed Chaibi (5): ASoC: dt-bindings: ti,twl4030-audio: convert to DT schema ASoC: dt-bindings: omap-twl4030: convert to DT schema ASoC: dt-bindings: everest,es8316: Document routing strings ASoC: dt-bindings: nuvoton,nau8825: Document routing strings ASoC: dt-bindings: wlf,wm8960: Document routing strings (pin names) Jonas Rebmann (1): ASoC: dt-bindings: asahi-kasei,ak4458: Reference common DAI properties Jyri Sarha (6): ASoC: SOF: sof-client-probes-ipc4: Implement ipc4_probes_points_info ASoC: SOF: IPC4: Add GET macros for module id and module instance id ASoC: SOF: sof-client: Add sof_client_ipc4_find_swidget_by_id() ASoC: SOF: sof-client-probes-ipc4: Human readable debugfs "probe_points" ASoC: SOF: sof-client-probes: Add available points_info(), IPC4 only ASoC: sof: ipc4-topology: Add support to sched_domain attribute Krzysztof Kozlowski (1): ASoC: dt-bindings: Minor whitespace cleanup in example Kuninori Morimoto (27): ASoC: soc-dapm: move snd_soc_dapm_get_bias_level() to soc-dpcm ASoC: soc-dapm: move snd_soc_dapm_init_bias_level() to soc-dpcm ASoC: soc-component: unpack snd_soc_component_force_bias_level() ASoC: soc-component: unpack snd_soc_component_get_bias_level() ASoC: soc-component: unpack snd_soc_component_init_bias_level() ASoC: renesas: msiof: msiof_update_and_wait() checks whether reg was updated ASoC: renesas: msiof: tidyup error message ASoC: renesas: msiof: cleanup status clear method ASoC: renesas: msiof: start DMAC first ASoC: soc-dapm: rename snd_soc_dapm_kcontrol_widget() to snd_soc_dapm_kcontrol_to_widget() ASoC: soc-dapm: rename snd_soc_dapm_kcontrol_dapm() to snd_soc_dapm_kcontrol_to_dapm() ASoC: soc-dapm: rename dapm_kcontrol_get_value() to snd_soc_dapm_kcontrol_get_value() ASoC: soc-dapm: rename snd_soc_kcontrol_component() to snd_soc_kcontrol_to_component() ASoC: renesas: msiof: Use guard() for spin locks ASoC: renesas: rsnd: Use guard() for spin locks ASoC: renesas: fsi: Use guard() for spin locks ASoC: renesas: rz-ssi: Use guard() for spin locks ASoC: soc-dapm: add snd_soc_dapm_to_dev() ASoC: soc-dapm: add snd_soc_dapm_to_card() ASoC: soc-dapm: use dapm->component instead of container_of() ASoC: soc-component: add snd_soc_component_to_dapm() ASoC: soc-card: add snd_soc_card_to_dapm() ASoC: soc-dapm: remove suspend_bias_off from snd_soc_dapm_context ASoC: soc-dapm: tidyup idle_bias handling - step1 ASoC: soc-dapm: tidyup idle_bias handling - step2 ASoC: soc-dapm: add snd_soc_dapm_get_bias_level() ASoC: soc-dapm: add snd_soc_dapm_set_idle_bias() Liao Yuanhong (1): ASoC: Intel: sst: Remove redundant semicolons Mahdi Khosravi (1): ASoC: dt-bindings: realtek,alc5623: convert to DT schema Marco Crivellari (1): ASoC: replace use of system_unbound_wq with system_dfl_wq Mark Brown (29): ASoC: unpack snd_soc_component_xxx_bias_level() ASoC: Intel: avs: Conditional path support ASoC: codecs: Add support for FourSemi FS2104/5S gpiolib: fence off legacy interfaces Handle shared reset GPIO for WSA883x speakers ASoC: es8323: power and mixer controls cleanup and ASoC: Intel: avs: Code cleanups and separation ASoC: qcom: audioreach: cleanup and calibration ASoC: dt-bindings: Convert TI TWL4030 sound ASoC: Use kcalloc() instead of kzalloc() ASoC: renesas: msiof: Make small adjustments to avoid ASoC: use int type to store negative error codes ASoC: Intel: avs: Refactor machine-board registration ASoC: SOF: Intel: only detect codecs when HDA DSP ASoC: SOF: sof-client: Enforce client access scope ASoC: SOF: sof-client-probes-ipc4: Query available ASoC: SOF: ipc4: Add support for 8-bit and float ASoC: qcom: audioreach: fix sparse warnings ASoC: Intel: avs: Adjust platform names ASoC: dt-bindings: Document routing strings for Improve cs42l43 suspend/IRQ interactions ASoC: Intel: Fix invalid quirk input mapping ASoC: renesas: Use guard() for spin locks ASoC: codecs: pcm1754: add pcm1754 dac driver Add PM4125 audio codec driver ASoC: cs35l56: Handle vendor-specific UEFI ASoC: soc-dapm: prepare for hiding struct ASoC: codecs: wcd93xxx: remove code duplication ASoC: fsl: fsl_qmc_audio: Reduce amount of Matthias Schiffer (1): ASoC: tlv320aic32x4: use dev_err_probe() for regulators Mohammad Rafi Shaik (3): ASoC: dt-bindings: qcom,wsa8830: Add reset-gpios for shared line ASoC: codecs: wsa883x: Handle shared reset GPIO for WSA883x speakers ASoC: dt-bindings: qcom,lpass-va-macro: Update bindings for clocks to support ADSP Nick Li (5): dt-bindings: vendor-prefixes: Add Shanghai FourSemi Semiconductor Co.,Ltd ASoC: dt-bindings: Add FS2104/5S audio amplifiers ASoC: codecs: Add library for FourSemi audio amplifiers ASoC: codecs: Add FourSemi FS2104/5S audio amplifier driver MAINTAINERS: Add entry for FourSemi audio amplifiers Peter Ujfalusi (3): ASoC: SOF: ipc4-pcm: Add fixup for channels ASoC: SOF: sof-client: Convert sof_client_dev_to_sof_dev into function ASoC: SOF: sof-client: Introduce sof_client_dev_entry structure Qianfeng Rong (6): ASoC: codecs: Use kcalloc() instead of kzalloc() ASoC: fsl: Use kcalloc() instead of kzalloc() ASoC: test-component: Use kcalloc() instead of kzalloc() ASoC: amd: use int type to store negative error codes ASoC: adau1977: use int type to store negative error codes ASoC: qcom: use int type to store negative error codes Richard Fitzgerald (6): ASoC: cs-amp-lib: Rename defines for Cirrus Logic EFI ASoC: cs-amp-lib: Add handling for Lenovo and HP UEFI speaker ID ASoC: cs35l56: Check for vendor-specific speaker ID value ASoC: cs-amp-lib-test: Add tests for cs_amp_get_vendor_spkid() ASoC: cs-amp-lib: Add HP-specific EFI variable for calibration data ASoC: cs-amp-lib-test: Add test for getting cal data from HP EFI Rob Herring (Arm) (3): ASoC: dt-bindings: Drop imx-audio-sgtl5000.txt ASoC: dt-bindings: Convert brcm,bcm2835-i2s to DT schema ASoC: dt-bindings: linux,spdif: Add "port" node Samasth Norway Ananda (1): ASoC: SOF: ipc3-dtrace: fix potential integer overflow in allocation Seppo Ingalsuo (2): ASoC: SOF: ipc4-topology: Add support for 8-bit formats ASoC: SOF: ipc4-topology: Add support for float sample type Shenghao Ding (1): ASoC: tas2781: Add keyword "init" in profile section Shengjiu Wang (1): ASoC: imx-hdmi: remove cpu_pdev related code Shimrra Shai (4): ASoC: es8323: enable DAPM power widgets for playback DAC and output ASoC: es8323: remove DAC enablement write from es8323_probe ASoC: es8323: add proper left/right mixer controls via DAPM ASoC: es8323: enable DAPM power widgets for playback DAC Shuming Fan (2): ASoC: rt721: add some vendor registers ASoC: rt1321: Add RT1321 amplifier support Srinivas Kandagatla (28): ASoC: qcom: audioreach: deprecate AR_TKN_U32_MODULE_[IN/OUT]_PORTS ASoC: qcom: audioreach: add documentation for i2s interface type ASoC: qcom: audioreach: add support for static calibration ASoC: qcom: audioreach: fix typos in I2S_INTF_TYPE ASoC: qcom: audioreach: sort modules based on hex ids ASoC: qcom: audioreach: add support for SMECNS module ASoC: qcom: audioreach: fix potential null pointer dereference ASoC: qcom: topology: convert to cpu endainess type before accessing ASoC: qcom: audioreach: convert to cpu endainess type before accessing ASoC: qcom: x1e80100: set card driver name from match data ASoC: dt-bindings: qcom,sm8250: Add glymur sound card ASoC: qcom: x1e80100: add compatible for glymur SoC ASoC: dt-bindings: qcom: Add Glymur LPASS wsa and va macro codecs ASoC: codecs: lpass-macro: add Codec version 2.9 ASoC: codecs: lpass-wsa-macro: add Codev version 2.9 ASoC: codecs: wcd937x: set the comp soundwire port correctly ASoC: codecs: wcd937x: make stub functions inline soundwire: bus: add of_sdw_find_device_by_node helper soundwire: bus: add sdw_slave_get_current_bank helper ASoC: codecs: wcdxxxx: use of_sdw_find_device_by_node helper ASoC: codecs: wcdxxxx: use sdw_slave_get_current_bank helper ASoC: codecs: wcd: add common helper for wcd codecs ASoC: codecs: wcd-common: move WCD_SDW_CH to common ASoC: codecs: wcd-common: move component ops to common ASoC: codecs: wcd939x: get regmap directly ASoC: codecs: wcd-common: move status_update callback to common ASoC: codecs: wcd938x: get regmap directly ASoC: codecs: wcd937x: get regmap directly Stefan Kerkmann (1): ASoC: dt-bindings: ti,pcm1754: add binding documentation Takashi Iwai (3): ASoC: Intel: bytcht_es8316: Fix invalid quirk input mapping ASoC: Intel: bytcr_rt5640: Fix invalid quirk input mapping ASoC: Intel: bytcr_rt5651: Fix invalid quirk input mapping Terry Cheong (1): ASoC: mediatek: Use SND_JACK_AVOUT for HDMI/DP jacks Waqar Hameed (1): ASoC: SOF: imx: Remove error print for devm_add_action_or_reset() Wolfram Sang (1): ASoC: remove unneeded 'fast_io' parameter in regmap_config Xichao Zhao (3): ASoC: cs48l32: Use PTR_ERR_OR_ZERO() to simplify code ASoC: fix "dependant"->"dependent" ASoC: SOF: imx: Remove the use of dev_err_probe() .../devicetree/bindings/mfd/twl4030-audio.txt | 46 - .../devicetree/bindings/sound/alc5623.txt | 25 - .../bindings/sound/asahi-kasei,ak4458.yaml | 4 + .../devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 24 - .../bindings/sound/brcm,bcm2835-i2s.yaml | 51 + .../devicetree/bindings/sound/everest,es8316.yaml | 16 + .../bindings/sound/foursemi,fs2105s.yaml | 101 ++ .../devicetree/bindings/sound/fsl,easrc.yaml | 2 +- .../devicetree/bindings/sound/fsl,imx-asrc.yaml | 2 +- .../bindings/sound/imx-audio-sgtl5000.txt | 56 - .../devicetree/bindings/sound/linux,spdif.yaml | 3 + .../devicetree/bindings/sound/nuvoton,nau8825.yaml | 14 + .../devicetree/bindings/sound/omap-twl4030.txt | 62 - .../bindings/sound/qcom,lpass-va-macro.yaml | 24 +- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 1 + .../bindings/sound/qcom,pm4125-codec.yaml | 134 ++ .../devicetree/bindings/sound/qcom,pm4125-sdw.yaml | 79 + .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + .../devicetree/bindings/sound/qcom,wsa883x.yaml | 11 +- .../devicetree/bindings/sound/realtek,alc5623.yaml | 54 + .../devicetree/bindings/sound/ti,omap-twl4030.yaml | 98 ++ .../devicetree/bindings/sound/ti,pcm1754.yaml | 55 + .../bindings/sound/ti,twl4030-audio.yaml | 90 + .../devicetree/bindings/sound/wlf,wm8960.yaml | 22 + .../devicetree/bindings/vendor-prefixes.yaml | 2 + Documentation/sound/soc/codec.rst | 4 +- Documentation/sound/soc/platform.rst | 4 +- MAINTAINERS | 10 + drivers/soc/fsl/qe/qmc.c | 44 +- drivers/soundwire/bus.c | 12 + drivers/soundwire/slave.c | 6 + include/linux/soundwire/sdw.h | 17 + include/sound/cs-amp-lib.h | 1 + include/sound/dmaengine_pcm.h | 5 + include/sound/soc-component.h | 79 +- include/sound/soc-dapm.h | 61 +- include/sound/soc.h | 5 + include/sound/sof/ipc4/header.h | 4 + include/sound/tas2781-dsp.h | 8 + include/sound/tlv320dac33-plat.h | 21 - include/uapi/sound/intel/avs/tokens.h | 15 + include/uapi/sound/snd_ar_tokens.h | 20 +- include/uapi/sound/sof/tokens.h | 2 + sound/core/pcm_dmaengine.c | 2 + sound/soc/amd/acp/acp-rembrandt.c | 2 +- sound/soc/amd/acp/amd-sdw-acpi.c | 2 +- sound/soc/amd/raven/acp3x-i2s.c | 3 +- sound/soc/amd/vangogh/acp5x-i2s.c | 3 +- sound/soc/codecs/Kconfig | 53 +- sound/soc/codecs/Makefile | 18 +- sound/soc/codecs/adau1977.c | 2 +- sound/soc/codecs/cs-amp-lib-test.c | 252 ++- sound/soc/codecs/cs-amp-lib.c | 139 +- sound/soc/codecs/cs35l56-shared.c | 12 +- sound/soc/codecs/cs42l43-jack.c | 8 +- sound/soc/codecs/cs42l43.c | 109 +- sound/soc/codecs/cs42l43.h | 3 + sound/soc/codecs/cs48l32-tables.c | 4 +- sound/soc/codecs/da7213.c | 6 +- sound/soc/codecs/es8323.c | 17 +- sound/soc/codecs/fs-amp-lib.c | 265 +++ sound/soc/codecs/fs-amp-lib.h | 150 ++ sound/soc/codecs/fs210x.c | 1586 +++++++++++++++++ sound/soc/codecs/fs210x.h | 75 + sound/soc/codecs/lpass-macro-common.h | 1 + sound/soc/codecs/lpass-va-macro.c | 2 + sound/soc/codecs/lpass-wsa-macro.c | 2 + sound/soc/codecs/pcm1754.c | 185 ++ sound/soc/codecs/pcm6240.c | 4 +- sound/soc/codecs/pm4125-sdw.c | 545 ++++++ sound/soc/codecs/pm4125.c | 1780 ++++++++++++++++++++ sound/soc/codecs/pm4125.h | 307 ++++ sound/soc/codecs/rt1320-sdw.c | 380 ++++- sound/soc/codecs/rt1320-sdw.h | 10 + sound/soc/codecs/rt721-sdca-sdw.c | 10 +- sound/soc/codecs/tas2781-fmwlib.c | 12 + sound/soc/codecs/tas2781-i2c.c | 6 + sound/soc/codecs/tlv320aic32x4.c | 18 +- sound/soc/codecs/tlv320dac33.c | 69 +- sound/soc/codecs/wcd-common.c | 143 ++ sound/soc/codecs/wcd-common.h | 46 + sound/soc/codecs/wcd934x.c | 82 +- sound/soc/codecs/wcd937x-sdw.c | 67 +- sound/soc/codecs/wcd937x.c | 93 +- sound/soc/codecs/wcd937x.h | 24 +- sound/soc/codecs/wcd938x-sdw.c | 83 +- sound/soc/codecs/wcd938x.c | 99 +- sound/soc/codecs/wcd938x.h | 26 +- sound/soc/codecs/wcd939x-sdw.c | 98 +- sound/soc/codecs/wcd939x.c | 104 +- sound/soc/codecs/wcd939x.h | 32 +- sound/soc/codecs/wl1273.c | 500 ------ sound/soc/codecs/wl1273.h | 16 - sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 6 +- sound/soc/codecs/wm8994.h | 12 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm_adsp.c | 2 +- sound/soc/codecs/wsa883x.c | 57 +- sound/soc/fsl/fsl_qmc_audio.c | 125 +- sound/soc/fsl/fsl_sai.c | 3 +- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-hdmi.c | 13 +- sound/soc/generic/test-component.c | 4 +- sound/soc/intel/atom/sst/sst.c | 2 +- sound/soc/intel/avs/apl.c | 1 + sound/soc/intel/avs/avs.h | 90 +- sound/soc/intel/avs/board_selection.c | 309 ++-- sound/soc/intel/avs/boards/da7219.c | 15 +- sound/soc/intel/avs/boards/dmic.c | 80 +- sound/soc/intel/avs/boards/es8336.c | 18 +- sound/soc/intel/avs/boards/hdaudio.c | 13 +- sound/soc/intel/avs/boards/i2s_test.c | 15 +- sound/soc/intel/avs/boards/max98357a.c | 15 +- sound/soc/intel/avs/boards/max98373.c | 15 +- sound/soc/intel/avs/boards/max98927.c | 15 +- sound/soc/intel/avs/boards/nau8825.c | 15 +- sound/soc/intel/avs/boards/probe.c | 49 +- sound/soc/intel/avs/boards/rt274.c | 17 +- sound/soc/intel/avs/boards/rt286.c | 15 +- sound/soc/intel/avs/boards/rt298.c | 15 +- sound/soc/intel/avs/boards/rt5514.c | 15 +- sound/soc/intel/avs/boards/rt5640.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 15 +- sound/soc/intel/avs/boards/rt5682.c | 15 +- sound/soc/intel/avs/boards/ssm4567.c | 15 +- sound/soc/intel/avs/cnl.c | 1 + sound/soc/intel/avs/core.c | 1 + sound/soc/intel/avs/debug.h | 91 + sound/soc/intel/avs/debugfs.c | 1 + sound/soc/intel/avs/icl.c | 1 + sound/soc/intel/avs/ipc.c | 1 + sound/soc/intel/avs/lnl.c | 1 + sound/soc/intel/avs/mtl.c | 1 + sound/soc/intel/avs/path.c | 267 +++ sound/soc/intel/avs/path.h | 13 + sound/soc/intel/avs/pcm.c | 35 +- sound/soc/intel/avs/probes.c | 23 +- sound/soc/intel/avs/ptl.c | 1 + sound/soc/intel/avs/skl.c | 1 + sound/soc/intel/avs/tgl.c | 1 + sound/soc/intel/avs/topology.c | 110 ++ sound/soc/intel/avs/topology.h | 8 + sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 22 +- sound/soc/intel/boards/bytcr_rt5640.c | 9 +- sound/soc/intel/boards/bytcr_rt5651.c | 28 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/sof_es8336.c | 2 +- sound/soc/mediatek/common/mtk-soundcard-driver.c | 19 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 2 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 8 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 2 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 4 +- sound/soc/pxa/Kconfig | 4 +- sound/soc/qcom/lpass-cdc-dma.c | 3 +- sound/soc/qcom/lpass-hdmi.c | 2 +- sound/soc/qcom/qdsp6/audioreach.c | 27 + sound/soc/qcom/qdsp6/audioreach.h | 18 +- sound/soc/qcom/qdsp6/topology.c | 45 +- sound/soc/qcom/sc8280xp.c | 3 + sound/soc/qcom/x1e80100.c | 5 +- sound/soc/renesas/fsi.c | 30 +- sound/soc/renesas/rcar/core.c | 18 +- sound/soc/renesas/rcar/msiof.c | 58 +- sound/soc/renesas/rcar/src.c | 19 +- sound/soc/renesas/rcar/ssi.c | 35 +- sound/soc/renesas/rz-ssi.c | 14 +- sound/soc/soc-core.c | 4 +- sound/soc/soc-dapm.c | 121 +- sound/soc/sof/imx/imx-common.c | 6 +- sound/soc/sof/imx/imx8.c | 3 +- sound/soc/sof/intel/hda-codec.c | 3 - sound/soc/sof/intel/hda-ctrl.c | 8 +- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda-sdw-bpt.c | 2 +- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 2 +- sound/soc/sof/ipc3-dtrace.c | 2 +- sound/soc/sof/ipc4-pcm.c | 56 + sound/soc/sof/ipc4-topology.c | 187 +- sound/soc/sof/ipc4-topology.h | 16 + sound/soc/sof/sof-audio.h | 3 + sound/soc/sof/sof-client-probes-ipc3.c | 25 +- sound/soc/sof/sof-client-probes-ipc4.c | 134 +- sound/soc/sof/sof-client-probes.c | 60 +- sound/soc/sof/sof-client-probes.h | 9 +- sound/soc/sof/sof-client.c | 118 +- sound/soc/sof/sof-client.h | 8 +- sound/soc/sof/sof-priv.h | 9 + 193 files changed, 8911 insertions(+), 2552 deletions(-) delete mode 100644 Documentation/devicetree/bindings/mfd/twl4030-audio.txt delete mode 100644 Documentation/devicetree/bindings/sound/alc5623.txt delete mode 100644 Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/foursemi,fs2105s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt delete mode 100644 Documentation/devicetree/bindings/sound/omap-twl4030.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,pm4125-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,pm4125-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,alc5623.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,omap-twl4030.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1754.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,twl4030-audio.yaml delete mode 100644 include/sound/tlv320dac33-plat.h create mode 100644 sound/soc/codecs/fs-amp-lib.c create mode 100644 sound/soc/codecs/fs-amp-lib.h create mode 100644 sound/soc/codecs/fs210x.c create mode 100644 sound/soc/codecs/fs210x.h create mode 100644 sound/soc/codecs/pcm1754.c create mode 100644 sound/soc/codecs/pm4125-sdw.c create mode 100644 sound/soc/codecs/pm4125.c create mode 100644 sound/soc/codecs/pm4125.h create mode 100644 sound/soc/codecs/wcd-common.c create mode 100644 sound/soc/codecs/wcd-common.h delete mode 100644 sound/soc/codecs/wl1273.c delete mode 100644 sound/soc/codecs/wl1273.h create mode 100644 sound/soc/intel/avs/debug.h

2 1

Qualcomm: Kaanapali: Add Kaanapali MTP HiFi config
by GitHub pull_request - opened 19 Sep '25

19 Sep '25

alsa-project/alsa-ucm-conf pull request #613 was opened from PrasadKumpatla: Add UCM2 configs for the Qualcomm Kaanapali-MTP Board. Tested on-board Speakers playback, on-board Microphones. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/613 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/613.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/realtek: Add quirk for HP Spectre 14t-ea100
by Kaden Berger 19 Sep '25

19 Sep '25

--- patch_realtek.c 2025-06-22 11:36:47.365528716 -0500 +++ patch_realtek.c.new 2025-09-18 13:59:13.610449218 -0500 @@ -10779,6 +10779,7 @@ SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x89da, "HP Spectre x360 14t-ea100", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), SND_PCI_QUIRK(0x103c, 0x89e7, "HP Elite x2 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a0f, "HP Pavilion 14-ec1xxx", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a20, "HP Laptop 15s-fq5xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2),

2 1

[GIT PULL] ASoC fixes for v6.17-rc6
by Mark Brown 18 Sep '25

18 Sep '25

The following changes since commit 76eeb9b8de9880ca38696b2fb56ac45ac0a25c6c: Linux 6.17-rc5 (2025-09-07 14:22:57 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-rc6 for you to fetch changes up to 1dd28fd86c3fa4e395031dd6f2ba920242107010: ASoC: rt5682s: Adjust SAR ADC button mode to fix noise issue (2025-09-17 12:38:59 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.17 A pile of fixes that accumilated over the past few -rcs, this is all driver specifics including a small pile of quirks for new systems. ---------------------------------------------------------------- Amadeusz Sławiński (1): ASoC: Intel: catpt: Expose correct bit depth to userspace Balamurugan C (1): ASoC: Intel: PTL: Add entry for HDMI-In capture support to non-I2S codec boards. Charles Keepax (6): ASoC: wm8940: Correct PLL rate rounding ASoC: wm8940: Correct typo in control name ASoC: wm8974: Correct PLL rate rounding ASoC: SDCA: Fix return value in sdca_regmap_mbq_size() ASoC: SDCA: Fix return value in detected_mode_handler() ASoC: SDCA: Reorder members of hide struct to remove holes Colin Ian King (1): ASoC: SOF: Intel: hda-stream: Fix incorrect variable used in error message Dan Carpenter (1): ASoC: codec: sma1307: Fix memory corruption in sma1307_setting_loaded() Daniel Baluta (1): ASoC: SOF: imx: Fix devm_ioremap_resource check Jack Yu (1): ASoC: rt5682s: Adjust SAR ADC button mode to fix noise issue Krzysztof Kozlowski (3): ASoC: qcom: q6apm-lpass-dais: Fix NULL pointer dereference if source graph failed ASoC: codecs: lpass-rx-macro: Fix playback quality distortion ASoC: codecs: lpass-wsa-macro: Fix speaker quality distortion Mac Chiang (1): ASoC: Intel: sof_sdw: use PRODUCT_FAMILY for Fatcat series Maciej Strozek (1): ASoC: SDCA: Add quirk for incorrect function types for 3 systems Mark Brown (3): Minor bug fixes for some older Wolfson devices More minor SDCA bug fixes Fix lpaif_type and DAI configuration for I2S Mohammad Rafi Shaik (4): ASoC: qcom: audioreach: Fix lpaif_type configuration for the I2S interface ASoC: qcom: q6apm-lpass-dais: Fix missing set_fmt DAI op for I2S ASoC: qcom: sc8280xp: Enable DAI format configuration for MI2S interfaces ASoC: qcom: sc8280xp: Fix sound card driver name match data for QCS8275 Shuming Fan (1): ASoC: rt712: avoid skipping the blind write Syed Saba Kareem (1): ASoC: amd: amd_sdw: Add quirks for some new Dell laptops Venkata Prasad Potturu (2): ASoC: amd: acp: Adjust pdm gain value ASoC: amd: acp: Fix incorrect retrival of acp_chip_info include/sound/sdca.h | 1 + include/sound/sdca_function.h | 21 ++++++++++++--------- sound/soc/amd/acp/acp-i2s.c | 11 +++++------ sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 ++++++++++++++++ sound/soc/amd/acp/amd.h | 2 +- sound/soc/codecs/lpass-rx-macro.c | 22 +++++++++++++++------- sound/soc/codecs/lpass-wsa-macro.c | 22 +++++++++++++++------- sound/soc/codecs/rt5682s.c | 17 +++++++++-------- sound/soc/codecs/rt712-sdca.c | 6 ++---- sound/soc/codecs/sma1307.c | 7 ++++--- sound/soc/codecs/wm8940.c | 9 +++++++-- sound/soc/codecs/wm8974.c | 8 ++++++-- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/intel/boards/sof_ssp_amp.c | 6 ++++++ sound/soc/intel/catpt/pcm.c | 23 +++++++++++++++++------ sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 6 ++++++ sound/soc/qcom/qdsp6/audioreach.c | 1 + sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 7 +++++-- sound/soc/qcom/sc8280xp.c | 6 +++++- sound/soc/sdca/sdca_device.c | 20 ++++++++++++++++++++ sound/soc/sdca/sdca_functions.c | 13 ++++++++----- sound/soc/sdca/sdca_interrupts.c | 2 +- sound/soc/sdca/sdca_regmap.c | 2 +- sound/soc/sof/imx/imx-common.c | 4 ++-- sound/soc/sof/intel/hda-stream.c | 2 +- 25 files changed, 167 insertions(+), 69 deletions(-)

2 1

[PATCH] ASoC: rt5682s: Adjust SAR ADC button mode to fix noise issue
by Jack Yu 18 Sep '25

18 Sep '25

Adjust register settings for SAR adc button detection mode to fix noise issue in headset. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 80b921695e7d..1d80a4b862e2 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -653,14 +653,15 @@ static void rt5682s_sar_power_mode(struct snd_soc_component *component, int mode switch (mode) { case SAR_PWR_SAVING: snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, - RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_DIS); + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_EN); snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, - RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK, - RT5682S_CTRL_MB1_REG | RT5682S_CTRL_MB2_REG); + RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK | + RT5682S_VREF_POW_MASK, RT5682S_CTRL_MB1_FSM | + RT5682S_CTRL_MB2_FSM | RT5682S_VREF_POW_FSM); snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | - RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + RT5682S_SAR_BUTDET_POW_NORM | RT5682S_SAR_SEL_MB1_2_MANU); usleep_range(5000, 5500); snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_BUTDET_MASK, RT5682S_SAR_BUTDET_EN); @@ -688,7 +689,7 @@ static void rt5682s_sar_power_mode(struct snd_soc_component *component, int mode snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | - RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + RT5682S_SAR_BUTDET_POW_NORM | RT5682S_SAR_SEL_MB1_2_MANU); break; default: dev_err(component->dev, "Invalid SAR Power mode: %d\n", mode); @@ -725,7 +726,7 @@ static void rt5682s_disable_push_button_irq(struct snd_soc_component *component) snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | - RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + RT5682S_SAR_BUTDET_POW_NORM | RT5682S_SAR_SEL_MB1_2_MANU); } /** @@ -786,7 +787,7 @@ static int rt5682s_headset_detect(struct snd_soc_component *component, int jack_ jack_type = SND_JACK_HEADSET; snd_soc_component_write(component, RT5682S_SAR_IL_CMD_3, 0x024c); snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, - RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_EN); + RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_DIS); snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_SEL_MB1_2_MASK, val << RT5682S_SAR_SEL_MB1_2_SFT); rt5682s_enable_push_button_irq(component); @@ -966,7 +967,7 @@ static int rt5682s_set_jack_detect(struct snd_soc_component *component, RT5682S_EMB_JD_MASK | RT5682S_DET_TYPE | RT5682S_POL_FAST_OFF_MASK | RT5682S_MIC_CAP_MASK, RT5682S_EMB_JD_EN | RT5682S_DET_TYPE | - RT5682S_POL_FAST_OFF_HIGH | RT5682S_MIC_CAP_HS); + RT5682S_POL_FAST_OFF_LOW | RT5682S_MIC_CAP_HS); regmap_update_bits(rt5682s->regmap, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_POW_MASK, RT5682S_SAR_POW_EN); regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, -- 2.34.1

3 2

[PATCH v2] ASoC: amd: acp: Fix incorrect retrival of acp_chip_info
by Venkata Prasad Potturu 16 Sep '25

16 Sep '25

Use dev_get_drvdata(dev->parent) instead of dev_get_platdata(dev) to correctly obtain acp_chip_info members in the acp I2S driver. Previously, some members were not updated properly due to incorrect data access, which could potentially lead to null pointer dereferences. This issue was missed in the earlier commit ("ASoC: amd: acp: Fix NULL pointer deref in acp_i2s_set_tdm_slot"), which only addressed set_tdm_slot(). This change ensures that all relevant functions correctly retrieve acp_chip_info, preventing further null pointer dereference issues. Fixes: e3933683b25e ("ASoC: amd: acp: Remove redundant acp_dev_data structure") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Changes since v1: - Update the commit description. sound/soc/amd/acp/acp-i2s.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 617690362ad7..4ba0a66981ea 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -73,7 +73,7 @@ static int acp_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct device *dev = cpu_dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); int mode; mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; @@ -199,7 +199,7 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ u32 reg_val, fmt_reg, tdm_fmt; u32 lrclk_div_val, bclk_div_val; - chip = dev_get_platdata(dev); + chip = dev_get_drvdata(dev->parent); rsrc = chip->rsrc; /* These values are as per Hardware Spec */ @@ -386,7 +386,7 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct { struct acp_stream *stream = substream->runtime->private_data; struct device *dev = dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_resource *rsrc = chip->rsrc; u32 val, period_bytes, reg_val, ier_val, water_val, buf_size, buf_reg; @@ -516,14 +516,13 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct device *dev = dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_resource *rsrc = chip->rsrc; struct acp_stream *stream = substream->runtime->private_data; u32 reg_dma_size = 0, reg_fifo_size = 0, reg_fifo_addr = 0; u32 phy_addr = 0, acp_fifo_addr = 0, ext_int_ctrl; unsigned int dir = substream->stream; - chip = dev_get_platdata(dev); switch (dai->driver->id) { case I2S_SP_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { @@ -632,7 +631,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d { struct acp_stream *stream = substream->runtime->private_data; struct device *dev = dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_resource *rsrc = chip->rsrc; unsigned int dir = substream->stream; unsigned int irq_bit = 0; -- 2.43.0

3 2

[PATCH] ASoC: amd: acp: Fix incorrect retrival of acp_chip_info
by Venkata Prasad Potturu 16 Sep '25

16 Sep '25

Use dev_get_drvdata(dev->parent) instead of dev_get_platdata(dev) to correctly get acp_chip_info members in acp I2S driver. This resolves issues where some members were zero due to incorrect data access. Fixes: e3933683b25e ("ASoC: amd: acp: Remove redundant acp_dev_data structure") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/acp-i2s.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 617690362ad7..4ba0a66981ea 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -73,7 +73,7 @@ static int acp_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct device *dev = cpu_dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); int mode; mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; @@ -199,7 +199,7 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ u32 reg_val, fmt_reg, tdm_fmt; u32 lrclk_div_val, bclk_div_val; - chip = dev_get_platdata(dev); + chip = dev_get_drvdata(dev->parent); rsrc = chip->rsrc; /* These values are as per Hardware Spec */ @@ -386,7 +386,7 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct { struct acp_stream *stream = substream->runtime->private_data; struct device *dev = dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_resource *rsrc = chip->rsrc; u32 val, period_bytes, reg_val, ier_val, water_val, buf_size, buf_reg; @@ -516,14 +516,13 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct device *dev = dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_resource *rsrc = chip->rsrc; struct acp_stream *stream = substream->runtime->private_data; u32 reg_dma_size = 0, reg_fifo_size = 0, reg_fifo_addr = 0; u32 phy_addr = 0, acp_fifo_addr = 0, ext_int_ctrl; unsigned int dir = substream->stream; - chip = dev_get_platdata(dev); switch (dai->driver->id) { case I2S_SP_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { @@ -632,7 +631,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d { struct acp_stream *stream = substream->runtime->private_data; struct device *dev = dai->component->dev; - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_resource *rsrc = chip->rsrc; unsigned int dir = substream->stream; unsigned int irq_bit = 0; -- 2.43.0

4 4

snd_tlv_convert_to_dB: Fix mute handling for MINMAX_MUTE type
by GitHub pull_request - opened 15 Sep '25

15 Sep '25

alsa-project/alsa-lib pull request #478 was opened from qaqland: Ensure the SND_CTL_TLV_DB_GAIN_MUTE value is returned when the calculated gain equals the minimum dB value for the SNDRV_CTL_TLVT_DB_MINMAX_MUTE type. The previous check based solely on the volume value could miss cases where the linear calculation resulted in the minimum gain. The db range of some devices is very small <img width="619" height="106" alt="17574693041385" src="https://github.com/user-attachments/assets/3461c643-284d-4ef0-8315-8fc68268…" /> Request URL : https://github.com/alsa-project/alsa-lib/pull/478 Patch URL : https://github.com/alsa-project/alsa-lib/pull/478.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Bug with USB Audio sound card (alsa-info provided)
by Nicholas Kreimeyer 14 Sep '25

14 Sep '25

Hello dear Alsa devs, I have an Arylic B50 SE USB (and BT) Audio interface for driving my speakers. It works fine except that the volume control seems to be bugged. With alsamixer I can set the volume of the device from 0-100 but the master level (when its main device) only changes from 50-100. All values below 50 are completely silent. I have provided alsa-info about my system here: https://alsa-project.org/db/?f=df0b4e88103e32de5be947419e734ba2ce55af1b The USB sound card is ID 3 "Arylic Up2Stream USB Audio at usb-0000:10:00.0-6, full speed" Dmesg output when connecting the device: [ 689.758406] usb 1-6: new full-speed USB device number 16 using xhci_hcd [ 690.028312] usb 1-6: not running at top speed; connect to a high speed hub [ 690.048287] usb 1-6: New USB device found, idVendor=8888, idProduct=1717, bcdDevice= 1.00 [ 690.048290] usb 1-6: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 690.048292] usb 1-6: Product: Up2Stream USB Audio [ 690.048293] usb 1-6: Manufacturer: Arylic [ 690.048294] usb 1-6: SerialNumber: 20190808 [ 690.132014] usb 1-6: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. [ 690.132018] usb 1-6: [2] FU [PCM Playback Volume] ch = 2, val = 0/4096/1 [ 690.141210] input: Arylic Up2Stream USB Audio as /devices/pci0000:00/0000:00:02.1/0000:05:00.0/0000:06:08.0/0000:08:00.0/0000:09:0c.0/0000:10:00.0/usb1/1-6/1-6:1.2/0003:8888:1717.0013/input/input33 [ 690.191459] hid-generic 0003:8888:1717.0013: input,hidraw13: USB HID v2.01 Device [Arylic Up2Stream USB Audio] on usb-0000:10:00.0-6/input2 [ 690.191615] usbhid 1-6:1.3: couldn't find an input interrupt endpoint If possible, can that bug with the volume controls be fixed? If it is a rather easy fix I can probably code it myself if given a few pointers of where to start. If you need additional info from me, please don't hesitate to ask. Thank you for your work on ALSA and with regards Nico Kreimeyer

1 0

[PATCH 2/2] ALSA: hda/realtek: Support Lenovo Thinkbook 13x Gen 5
by Bou-Saan Che 14 Sep '25

14 Sep '25

The laptop does not contain valid _DSD for these amps, so requires entries into the CS35L41 configuration table to function correctly. Signed-off-by: Bou-Saan Che <yungmeat(a)inboxia.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2627e2f49316..36dc0389c9bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11368,6 +11368,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3913, "Lenovo 145", ALC236_FIXUP_LENOVO_INV_DMIC), SND_PCI_QUIRK(0x17aa, 0x391f, "Yoga S990-16 pro Quad YC Quad", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3920, "Yoga S990-16 pro Quad VECO Quad", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3929, "Thinkbook 13x Gen 5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x392b, "Thinkbook 13x Gen 5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), -- 2.51.0

1 0

[PATCH] ALSA: hda/realtek: Fix volume adjustment issue on Lenovo Thinkbook 13x Gen 4
by Bou-Saan Che 14 Sep '25

14 Sep '25

This patch fixes the volume adjustment issue on this laptop by applying the necessary quirk configuration for the Realtek ALC287 codec. The issue was caused by incorrect configuration in the driver, which prevented proper volume control on certain systems. Signed-off-by: Bou-Saan Che <yungmeat(a)inboxia.org> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2627e2f49316..1981e3374b3e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11346,8 +11346,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { 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, 0x38c7, "Thinkbook 13x Gen 4", ALC287_FIXUP_CS35L41_I2C_4), - SND_PCI_QUIRK(0x17aa, 0x38c8, "Thinkbook 13x Gen 4", ALC287_FIXUP_CS35L41_I2C_4), + SND_PCI_QUIRK(0x17aa, 0x38c7, "Thinkbook 13x Gen 4", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38c8, "Thinkbook 13x Gen 4", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 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, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), -- 2.51.0

1 0

[PATCH 1/2] ALSA: hda: cs35l41: Support Lenovo Thinkbook 13x Gen 5
by Bou-Saan Che 14 Sep '25

14 Sep '25

This laptop does not contain _DSD so needs to be supported using the configuration table. Signed-off-by: Bou-Saan Che <yungmeat(a)inboxia.org> --- sound/pci/hda/cs35l41_hda_property.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index d8249d997c2a..16d5ea77192f 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -135,6 +135,8 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "17AA38C8", 4, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT }, 0, 2, -1, 1000, 4500, 24 }, { "17AA38F9", 2, EXTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38FA", 2, EXTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, + { "17AA3929", 4, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT }, 0, 2, -1, 1000, 4500, 24 }, + { "17AA392B", 4, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT }, 0, 2, -1, 1000, 4500, 24 }, {} }; @@ -558,6 +560,8 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "17AA38C8", generic_dsd_config }, { "CSC3551", "17AA38F9", generic_dsd_config }, { "CSC3551", "17AA38FA", generic_dsd_config }, + { "CSC3551", "17AA3929", generic_dsd_config }, + { "CSC3551", "17AA392B", generic_dsd_config }, {} }; -- 2.51.0

1 0

Internal microphone (Conexant SN6140) nonfunctional on Lenovo IdeaPad Slim 5 16AKP10 83HY after alsa-ucm-conf 1.2.14+
by GitHub issues - edited 12 Sep '25

12 Sep '25

alsa-project/alsa-ucm-conf issue #612 was edited from JayJaber: The internal microphone on a Lenovo IdeaPad Slim 5 16AKP10 83HY with Conexant SN6140 codec behaves as follows: On alsa-ucm-conf 1.2.14 and newer: - Family 17h/19h/1ah HD Audio Controller Digital Microphone appears as a recording device but does not record. Only a small bit of static is audible at the start and end of recordings. - Family 17h/19h/1ah HD Audio Controller Stereo Microphone appears as inactive/unplugged. When headphones are plugged in, it becomes active and the headphone microphone works under that device. After downgrading to alsa-ucm-conf 1.2.13-2: - Both Family 17h/19h/1ah HD Audio Controller Internal Stereo Microphone and Family 17h/19h/1ah HD Audio Controller Digital Microphone appear as active recording devices without headphones plugged in. - Digital Microphone still does not record usable audio, only a small bit of static at the beginning and end. - Internal Stereo Microphone works correctly for the built-in mic when headphones are not plugged in. Plugging in headphones causes the headphone microphone to be used under the Internal Stereo Microphone device. Summary: The Digital Microphone is detected but never records usable audio. On 1.2.13-2, the Internal Stereo Microphone routes audio correctly from the built-in mic, so recording works, but the Digital Microphone itself remains nonfunctional. On 1.2.14+: - Digital Microphone appears but remains completely nonfunctional. - Stereo Microphone shows as inactive/unplugged until headphones are connected, making the internal microphone unusable. - Headphone microphone routing works correctly. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/612 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Internal microphone (Conexant SN6140) detected but shows inactive/unplugged on Lenovo IdeaPad Slim 5 16AKP10 83HY after alsa-ucm-conf 1.2.14
by GitHub issues - edited 12 Sep '25

12 Sep '25

alsa-project/alsa-ucm-conf issue #612 was edited from JayJaber: The internal microphone on a Lenovo IdeaPad Slim 5 16AKP10 83HY with Conexant SN6140 codec behaves as follows: On alsa-ucm-conf 1.2.14 and newer: - Family 17h/19h/1ah HD Audio Controller Digital Microphone appears as a recording device but does not record. Only a small bit of static is audible at the start and end of recordings. - Family 17h/19h/1ah HD Audio Controller Stereo Microphone appears as inactive/unplugged. When headphones are plugged in, it becomes active and the headphone microphone works under that device. After downgrading to alsa-ucm-conf 1.2.13-2: - Both Family 17h/19h/1ah HD Audio Controller Internal Stereo Microphone and Family 17h/19h/1ah HD Audio Controller Digital Microphone appear as active recording devices without headphones plugged in. - Digital Microphone still does not record usable audio, only a small bit of static at the beginning and end. - Internal Stereo Microphone works correctly for the built-in mic when headphones are not plugged in. Plugging in headphones causes the headphone microphone to be used under the Internal Stereo Microphone device. Summary: The Digital Microphone is detected but never records usable audio. On 1.2.13-2, the Internal Stereo Microphone routes audio correctly from the built-in mic, so recording works, but the Digital Microphone itself remains nonfunctional. On 1.2.14+: - Digital Microphone appears but remains completely nonfunctional. - Stereo Microphone shows as inactive/unplugged until headphones are connected, making the internal microphone unusable. - Headphone microphone routing works correctly. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/612 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v5 1/5] ASoC: ops: improve snd_soc_get_volsw
by Niranjan H Y 12 Sep '25

12 Sep '25

* add error handling in case register read fails * clamp the values if the register value read is out of range Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> -- v5: - remove clamp parameter - move the boundary check after sign-bit extension --- sound/soc/soc-ops.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a629e0eac..03f74c32d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -118,6 +118,7 @@ static int soc_mixer_reg_to_ctl(struct soc_mixer_control *mc, unsigned int reg_v if (mc->sign_bit) val = sign_extend32(val, mc->sign_bit); + val = clamp(val, mc->min, mc->max); val -= mc->min; if (mc->invert) @@ -249,9 +250,15 @@ static int soc_get_volsw(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); unsigned int reg_val; - int val; + int val, ret = 0; reg_val = snd_soc_component_read(component, mc->reg); + val = reg_val; + if (val < 0) { + ret = val; + goto get_volsw_done; + } + val = soc_mixer_reg_to_ctl(mc, reg_val, mask, mc->shift, max); ucontrol->value.integer.value[0] = val; @@ -261,13 +268,19 @@ static int soc_get_volsw(struct snd_kcontrol *kcontrol, val = soc_mixer_reg_to_ctl(mc, reg_val, mask, mc->rshift, max); } else { reg_val = snd_soc_component_read(component, mc->rreg); + val = reg_val; + if (val < 0) { + ret = val; + goto get_volsw_done; + } val = soc_mixer_reg_to_ctl(mc, reg_val, mask, mc->shift, max); } ucontrol->value.integer.value[1] = val; } - return 0; +get_volsw_done: + return ret; } /** -- 2.45.2

2 6

Re: [PATCH v4 2/6] ASoC: ops: add snd_soc_get_volsw_clamped
by Holalu Yogendra, Niranjan 12 Sep '25

12 Sep '25

> From: Mark Brown <broonie(a)kernel.org> > Sent: Friday, September 12, 2025 12:02 AM > Subject: Re: [EXTERNAL] Re: [PATCH v4 2/6] ASoC: ops: add > snd_soc_get_volsw_clamped > > On Thu, Sep 11, 2025 at 05:21:14PM +0000, Holalu Yogendra, Niranjan wrote: > > > From: Mark Brown <broonie(a)kernel.org> > > > Since few of the drivers as still using the original API, and was not sure if > > everyone wanted to clamp the register value, did not want to disturb the > > original API snd_soc_get_volsw. Per my analysis, if register read fails due to > In what situation would anything want to report an invalid value? I am not sure if I fully follow. But in my case, for one of the cases, valid amp volume is 0x0 to 0x14 with mask 0x1f - so if the register value read is 0x15 to 0x1f, it is invalid. But in the next patch, I will add the logic to clamp the value. > > Are you thinking that this patch could be dropped and clamp the value to max in > > 'soc_get_volsw' (as in 1st patch) and use the same API ? Please suggest. > Yes, everything should report valid values. I/O errors should be > propagated as errors. I will update the next patch accordingly. Thanks Niranjan H Y

1 0

Lenovo Yoga 7 2-in-1 16AKP10 volume control bug
by Sandro Gähler 12 Sep '25

12 Sep '25

Hello I would like to report a bug for the brand new Lenovo Yoga 7 2-in-1 16AKP10. It has the same sound chip and also the same volume control bug as several older models. I thus assume that the same workaround, written by Philipp Jungkamp and later by Pascal Gross, will also work for this model. There are several lines with laptops that apply this workaround, for example this line in linux/sound/pci/hda/patch_realtek.c: SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN) Could you please add this line for my model? SND_PCI_QUIRK(0x????, 0x????, "Lenovo Yoga 7 2-in-1 16AKP10", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), Please advise on how to determine the first two parameters. Thanks a lot for your great work and support. Best regards, --Sandro Gähler

1 1

[PATCH v4 1/6] ASoC: ops: improve snd_soc_get_volsw
by Niranjan H Y 11 Sep '25

11 Sep '25

* add error handling in case register read fails * add support for clamping the values if the register value read is greater than max value Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- sound/soc/soc-ops.c | 38 +++++++++++++++++++++++++++++--------- 1 file changed, 29 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a629e0eac..59e91741b 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -111,10 +111,14 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); static int soc_mixer_reg_to_ctl(struct soc_mixer_control *mc, unsigned int reg_val, - unsigned int mask, unsigned int shift, int max) + unsigned int mask, unsigned int shift, int max, + bool clamp) { int val = (reg_val >> shift) & mask; + if (clamp) + val = clamp(val, 0, mc->max); + if (mc->sign_bit) val = sign_extend32(val, mc->sign_bit); @@ -245,29 +249,44 @@ static int soc_put_volsw(struct snd_kcontrol *kcontrol, static int soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol, - struct soc_mixer_control *mc, int mask, int max) + struct soc_mixer_control *mc, int mask, int max, + bool clamp) { struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); unsigned int reg_val; - int val; + int val, ret = 0; reg_val = snd_soc_component_read(component, mc->reg); - val = soc_mixer_reg_to_ctl(mc, reg_val, mask, mc->shift, max); + val = reg_val; + if (val < 0) { + ret = val; + goto get_volsw_done; + } + + val = soc_mixer_reg_to_ctl(mc, reg_val, mask, mc->shift, max, clamp); ucontrol->value.integer.value[0] = val; if (snd_soc_volsw_is_stereo(mc)) { if (mc->reg == mc->rreg) { - val = soc_mixer_reg_to_ctl(mc, reg_val, mask, mc->rshift, max); + val = soc_mixer_reg_to_ctl(mc, reg_val, mask, mc->rshift, + max, clamp); } else { reg_val = snd_soc_component_read(component, mc->rreg); - val = soc_mixer_reg_to_ctl(mc, reg_val, mask, mc->shift, max); + val = reg_val; + if (val < 0) { + ret = val; + goto get_volsw_done; + } + val = soc_mixer_reg_to_ctl(mc, reg_val, mask, + mc->shift, max, clamp); } ucontrol->value.integer.value[1] = val; } - return 0; +get_volsw_done: + return ret; } /** @@ -330,7 +349,8 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; unsigned int mask = soc_mixer_mask(mc); - return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max - mc->min); + return soc_get_volsw(kcontrol, ucontrol, mc, mask, + mc->max - mc->min, false); } EXPORT_SYMBOL_GPL(snd_soc_get_volsw); @@ -372,7 +392,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; unsigned int mask = soc_mixer_sx_mask(mc); - return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max); + return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max, false); } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); -- 2.45.2

3 7

[PATCH v1] ALSA: hda/tas2781: Fix a potential race condition that causes a NULL pointer in case no efi.get_variable exsits
by Shenghao Ding 11 Sep '25

11 Sep '25

A a potential race condition reported by one of my customers that leads to a NULL pointer dereference, where the call to efi.get_variable should be guarded with efi_rt_services_supported() to ensure that function exists. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/hda/codecs/side-codecs/tas2781_hda.c | 5 +++++ sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 5 +++++ 2 files changed, 10 insertions(+) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index 536940c78f00..96e6d82dc69e 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -193,6 +193,11 @@ int tas2781_save_calibration(struct tas2781_hda *hda) efi_status_t status; int i; + if (!efi_rt_services_supported(EFI_RT_SUPPORTED_GET_VARIABLE)) { + dev_err(p->dev, "%s: NO EFI FOUND!\n", __func__); + return -EINVAL; + } + if (hda->catlog_id < LENOVO) efi_guid = tasdev_fct_efi_guid[hda->catlog_id]; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 008dbe1490a7..4dea442d8c30 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -317,6 +317,11 @@ static int tas2563_save_calibration(struct tas2781_hda *h) unsigned int attr; int ret, i, j, k; + if (!efi_rt_services_supported(EFI_RT_SUPPORTED_GET_VARIABLE)) { + dev_err(p->dev, "%s: NO EFI FOUND!\n", __func__); + return -EINVAL; + } + cd->cali_dat_sz_per_dev = TAS2563_CAL_DATA_SIZE * TASDEV_CALIB_N; /* extra byte for each device is the device number */ -- 2.43.0

2 1

Internal microphone (Conexant SN6140) detected but shows inactive/unplugged on Lenovo IdeaPad Slim 5 16AKP10 83HY after alsa-ucm-conf 1.2.14
by GitHub issues - edited 11 Sep '25

11 Sep '25

alsa-project/alsa-ucm-conf issue #612 was edited from JayJaber: On a Lenovo IdeaPad Slim 5 16AKP10 83HY with Conexant SN6140 codec, the internal microphone works correctly with alsa-ucm-conf 1.2.13-2, but with 1.2.14 and newer it is detected but shows up as inactive/unplugged. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/612 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Internal microphone (Conexant SN6140) detected but shows inactive/unplugged on Lenovo IdeaPad Slim 5 16AKP10 83HY after alsa-ucm-conf 1.2.14
by GitHub issues - edited 11 Sep '25

11 Sep '25

alsa-project/alsa-ucm-conf issue #612 was edited from JayJaber: On a Lenovo IdeaPad Slim 5 16AKP10 83HY with Conexant SN6140 codec, the internal microphone works correctly with alsa-ucm-conf 1.2.13-2, but with 1.2.14 and newer it is not detected and shows up as inactive/unplugged. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/612 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Internal microphone (Conexant SN6140) not detected on Lenovo IdeaPad Slim 5 16AKP10 83HY after alsa-ucm-conf 1.2.14
by GitHub issues - opened 11 Sep '25

11 Sep '25

alsa-project/alsa-ucm-conf issue #612 was opened from JayJaber: On a Lenovo IdeaPad Slim 5 16AKP10 83HY with Conexant SN6140 codec, the internal microphone works correctly with alsa-ucm-conf 1.2.13-2, but with 1.2.14 and newer it is not detected and shows up as inactive/unplugged. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/612 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[tiwai-sound:test/auto-cleanup 208/210] sound/sparc/amd7930.c:357:45: error: 'flags' undeclared
by kernel test robot 10 Sep '25

10 Sep '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/auto-cleanup head: f29244cff0185975fee4f55293cca61cfd83af3e commit: db8ed7441c876cac679db2869b87b26633c89bde [208/210] ALSA: sparc/amd7930: Use guard() for spin locks config: sparc64-randconfig-002-20250910 (https://download.01.org/0day-ci/archive/20250911/202509110130.PlGyYo4k-lkp@…) compiler: sparc64-linux-gcc (GCC) 12.5.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250911/202509110130.PlGyYo4k-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/202509110130.PlGyYo4k-lkp@intel.com/ All errors (new ones prefixed by >>): sound/sparc/amd7930.c: In function 'amd7930_enable_ints': >> sound/sparc/amd7930.c:357:45: error: 'flags' undeclared (first use in this function) 357 | guard(spinlock_irqsave)(&amd->lock, flags); | ^~~~~ sound/sparc/amd7930.c:357:45: note: each undeclared identifier is reported only once for each function it appears in In file included from include/linux/module.h:17, from sound/sparc/amd7930.c:33: >> include/linux/cleanup.h:291:17: error: too many arguments to function 'class_spinlock_irqsave_constructor' 291 | class_##_name##_constructor | ^~~~~~ include/linux/cleanup.h:401:9: note: in expansion of macro 'CLASS' 401 | CLASS(_name, __UNIQUE_ID(guard)) | ^~~~~ sound/sparc/amd7930.c:357:9: note: in expansion of macro 'guard' 357 | guard(spinlock_irqsave)(&amd->lock, flags); | ^~~~~ include/linux/cleanup.h:477:33: note: declared here 477 | static inline class_##_name##_t class_##_name##_constructor(_type *l) \ | ^~~~~~ include/linux/cleanup.h:496:1: note: in expansion of macro '__DEFINE_LOCK_GUARD_1' 496 | __DEFINE_LOCK_GUARD_1(_name, _type, _lock) | ^~~~~~~~~~~~~~~~~~~~~ include/linux/spinlock.h:585:1: note: in expansion of macro 'DEFINE_LOCK_GUARD_1' 585 | DEFINE_LOCK_GUARD_1(spinlock_irqsave, spinlock_t, | ^~~~~~~~~~~~~~~~~~~ sound/sparc/amd7930.c: In function 'amd7930_disable_ints': sound/sparc/amd7930.c:365:45: error: 'flags' undeclared (first use in this function) 365 | guard(spinlock_irqsave)(&amd->lock, flags); | ^~~~~ >> include/linux/cleanup.h:291:17: error: too many arguments to function 'class_spinlock_irqsave_constructor' 291 | class_##_name##_constructor | ^~~~~~ include/linux/cleanup.h:401:9: note: in expansion of macro 'CLASS' 401 | CLASS(_name, __UNIQUE_ID(guard)) | ^~~~~ sound/sparc/amd7930.c:365:9: note: in expansion of macro 'guard' 365 | guard(spinlock_irqsave)(&amd->lock, flags); | ^~~~~ include/linux/cleanup.h:477:33: note: declared here 477 | static inline class_##_name##_t class_##_name##_constructor(_type *l) \ | ^~~~~~ include/linux/cleanup.h:496:1: note: in expansion of macro '__DEFINE_LOCK_GUARD_1' 496 | __DEFINE_LOCK_GUARD_1(_name, _type, _lock) | ^~~~~~~~~~~~~~~~~~~~~ include/linux/spinlock.h:585:1: note: in expansion of macro 'DEFINE_LOCK_GUARD_1' 585 | DEFINE_LOCK_GUARD_1(spinlock_irqsave, spinlock_t, | ^~~~~~~~~~~~~~~~~~~ -- amd7930.c: In function 'amd7930_enable_ints': amd7930.c:357:45: error: 'flags' undeclared (first use in this function) 357 | guard(spinlock_irqsave)(&amd->lock, flags); | ^~~~~ amd7930.c:357:45: note: each undeclared identifier is reported only once for each function it appears in In file included from include/linux/module.h:17, from amd7930.c:33: >> include/linux/cleanup.h:291:17: error: too many arguments to function 'class_spinlock_irqsave_constructor' 291 | class_##_name##_constructor | ^~~~~~ include/linux/cleanup.h:401:9: note: in expansion of macro 'CLASS' 401 | CLASS(_name, __UNIQUE_ID(guard)) | ^~~~~ amd7930.c:357:9: note: in expansion of macro 'guard' 357 | guard(spinlock_irqsave)(&amd->lock, flags); | ^~~~~ include/linux/cleanup.h:477:33: note: declared here 477 | static inline class_##_name##_t class_##_name##_constructor(_type *l) \ | ^~~~~~ include/linux/cleanup.h:496:1: note: in expansion of macro '__DEFINE_LOCK_GUARD_1' 496 | __DEFINE_LOCK_GUARD_1(_name, _type, _lock) | ^~~~~~~~~~~~~~~~~~~~~ include/linux/spinlock.h:585:1: note: in expansion of macro 'DEFINE_LOCK_GUARD_1' 585 | DEFINE_LOCK_GUARD_1(spinlock_irqsave, spinlock_t, | ^~~~~~~~~~~~~~~~~~~ amd7930.c: In function 'amd7930_disable_ints': amd7930.c:365:45: error: 'flags' undeclared (first use in this function) 365 | guard(spinlock_irqsave)(&amd->lock, flags); | ^~~~~ >> include/linux/cleanup.h:291:17: error: too many arguments to function 'class_spinlock_irqsave_constructor' 291 | class_##_name##_constructor | ^~~~~~ include/linux/cleanup.h:401:9: note: in expansion of macro 'CLASS' 401 | CLASS(_name, __UNIQUE_ID(guard)) | ^~~~~ amd7930.c:365:9: note: in expansion of macro 'guard' 365 | guard(spinlock_irqsave)(&amd->lock, flags); | ^~~~~ include/linux/cleanup.h:477:33: note: declared here 477 | static inline class_##_name##_t class_##_name##_constructor(_type *l) \ | ^~~~~~ include/linux/cleanup.h:496:1: note: in expansion of macro '__DEFINE_LOCK_GUARD_1' 496 | __DEFINE_LOCK_GUARD_1(_name, _type, _lock) | ^~~~~~~~~~~~~~~~~~~~~ include/linux/spinlock.h:585:1: note: in expansion of macro 'DEFINE_LOCK_GUARD_1' 585 | DEFINE_LOCK_GUARD_1(spinlock_irqsave, spinlock_t, | ^~~~~~~~~~~~~~~~~~~ vim +/flags +357 sound/sparc/amd7930.c 353 354 /* Enable chip interrupts. The amd->lock is not held. */ 355 static __inline__ void amd7930_enable_ints(struct snd_amd7930 *amd) 356 { > 357 guard(spinlock_irqsave)(&amd->lock, flags); 358 sbus_writeb(AMR_INIT, amd->regs + AMD7930_CR); 359 sbus_writeb(AM_INIT_ACTIVE, amd->regs + AMD7930_DR); 360 } 361 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v3 1/4] ASoc: tas2783A: Add soundwire based codec driver
by Niranjan H Y 10 Sep '25

10 Sep '25

TAS2783 is mono digital input class-D Smart Amplifier based on MIPI Alliance Soundwire interface. The driver supports loading the algorithm coefficients for one or more tas2783 chips. Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v3: - fix compilation issue for ARCH=hexagon clang compiler - Kconfig - use depends on for SOUNDWIRE and EFI instead of select - add REGMAP_SOUNDWIRE_MBQ v2: - append Volume to amp gain mixer control - remove unwanted debug log --- sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 1417 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 107 +++ 4 files changed, 1540 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b8d58d2fe..a7d25ab7a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -266,6 +266,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2770 imply SND_SOC_TAS2780 imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783_SDW imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -2074,6 +2075,19 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783_SDW + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP_SOUNDWIRE + select REGMAP_SOUNDWIRE_MBQ + select CRC32 + help + Enable support for Texas Instruments TAS2783A Digital input + mono Class-D and DSP-inside audio power amplifiers. TAS2783 + driver implements a flexible and configurable algorithm + cofficient setting, for one, two or multiple TAS2783 chips. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a476d6c45..1920fa17b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -316,6 +316,7 @@ snd-soc-tas2781-comlib-y := tas2781-comlib.o snd-soc-tas2781-comlib-i2c-y := tas2781-comlib-i2c.o snd-soc-tas2781-fmwlib-y := tas2781-fmwlib.o snd-soc-tas2781-i2c-y := tas2781-i2c.o +snd-soc-tas2783-sdw-y := tas2783-sdw.o snd-soc-tfa9879-y := tfa9879.o snd-soc-tfa989x-y := tfa989x.o snd-soc-tlv320adc3xxx-y := tlv320adc3xxx.o @@ -732,6 +733,7 @@ obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB_I2C) += snd-soc-tas2781-comlib-i2c.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783_SDW) += snd-soc-tas2783-sdw.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000..ddadfb7c5 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,1417 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2783 driver implements a flexible and configurable +// algo coefficient setting for single TAS2783 chips. +// +// Author: Niranjan H Y <niranjanhy(a)ti.com> +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> + +#include <linux/unaligned.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/wait.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +#define TIMEOUT_FW_DL_MS (3000) +#define FW_DL_OFFSET 36 +#define FW_FL_HDR 12 +#define TAS2783_PROBE_TIMEOUT 5000 +#define TAS2783_CALI_GUID EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, \ + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92) + +#if !IS_ENABLED(CONFIG_SND_SOC_TAS2783_UTIL) +static void tas25xx_register_misc(struct sdw_slave *peripheral) {} +static void tas25xx_deregister_misc(void) {} +#endif + +static const u32 tas2783_cali_reg[] = { + TAS2783_CAL_R0, + TAS2783_CAL_INVR0, + TAS2783_CAL_R0LOW, + TAS2783_CAL_POWER, + TAS2783_CAL_TLIM, +}; + +struct bin_header_t { + u16 vendor_id; + u16 version; + u32 file_id; + u32 length; +}; + +struct calibration_data { + u32 is_valid; + unsigned long read_sz; + u8 data[TAS2783_CALIB_DATA_SZ]; +}; + +struct tas2783_prv { + struct snd_soc_component *component; + struct calibration_data cali_data; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + /* calibration */ + struct mutex calib_lock; + /* pde and firmware download */ + struct mutex pde_lock; + struct regmap *regmap; + struct device *dev; + struct class *class; + struct attribute_group *cal_attr_groups; + struct tm tm; + u8 rca_binaryname[64]; + u8 dev_name[32]; + bool hw_init; + /* wq for firmware download */ + wait_queue_head_t fw_wait; + bool fw_dl_task_done; + bool fw_dl_success; +}; + +static const struct reg_default tas2783_reg_default[] = { + {TAS2783_AMP_LEVEL, 0x28}, + {TASDEV_REG_SDW(0, 0, 0x03), 0x28}, + {TASDEV_REG_SDW(0, 0, 0x04), 0x21}, + {TASDEV_REG_SDW(0, 0, 0x05), 0x41}, + {TASDEV_REG_SDW(0, 0, 0x06), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x07), 0x20}, + {TASDEV_REG_SDW(0, 0, 0x08), 0x09}, + {TASDEV_REG_SDW(0, 0, 0x09), 0x02}, + {TASDEV_REG_SDW(0, 0, 0x0a), 0x0a}, + {TASDEV_REG_SDW(0, 0, 0x0c), 0x10}, + {TASDEV_REG_SDW(0, 0, 0x0d), 0x13}, + {TASDEV_REG_SDW(0, 0, 0x0e), 0xc2}, + {TASDEV_REG_SDW(0, 0, 0x0f), 0x40}, + {TASDEV_REG_SDW(0, 0, 0x10), 0x04}, + {TASDEV_REG_SDW(0, 0, 0x13), 0x13}, + {TASDEV_REG_SDW(0, 0, 0x14), 0x12}, + {TASDEV_REG_SDW(0, 0, 0x15), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x16), 0x12}, + {TASDEV_REG_SDW(0, 0, 0x17), 0x80}, + {TAS2783_DVC_LVL, 0x00}, + {TASDEV_REG_SDW(0, 0, 0x1b), 0x61}, + {TASDEV_REG_SDW(0, 0, 0x1c), 0x36}, + {TASDEV_REG_SDW(0, 0, 0x1d), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x1f), 0x01}, + {TASDEV_REG_SDW(0, 0, 0x20), 0x2e}, + {TASDEV_REG_SDW(0, 0, 0x21), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x34), 0x06}, + {TASDEV_REG_SDW(0, 0, 0x35), 0xbd}, + {TASDEV_REG_SDW(0, 0, 0x36), 0xad}, + {TASDEV_REG_SDW(0, 0, 0x37), 0xa8}, + {TASDEV_REG_SDW(0, 0, 0x38), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x3b), 0xfc}, + {TASDEV_REG_SDW(0, 0, 0x3d), 0xdd}, + {TASDEV_REG_SDW(0, 0, 0x40), 0xf6}, + {TASDEV_REG_SDW(0, 0, 0x41), 0x14}, + {TASDEV_REG_SDW(0, 0, 0x5c), 0x19}, + {TASDEV_REG_SDW(0, 0, 0x5d), 0x80}, + {TASDEV_REG_SDW(0, 0, 0x63), 0x48}, + {TASDEV_REG_SDW(0, 0, 0x65), 0x08}, + {TASDEV_REG_SDW(0, 0, 0x66), 0xb2}, + {TASDEV_REG_SDW(0, 0, 0x67), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x6a), 0x12}, + {TASDEV_REG_SDW(0, 0, 0x6b), 0xfb}, + {TASDEV_REG_SDW(0, 0, 0x6c), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x6d), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x6e), 0x1a}, + {TASDEV_REG_SDW(0, 0, 0x6f), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x70), 0x96}, + {TASDEV_REG_SDW(0, 0, 0x71), 0x02}, + {TASDEV_REG_SDW(0, 0, 0x73), 0x08}, + {TASDEV_REG_SDW(0, 0, 0x75), 0xe0}, + {TASDEV_REG_SDW(0, 0, 0x7a), 0x60}, + {TASDEV_REG_SDW(0, 0, 0x60), 0x21}, + {TASDEV_REG_SDW(0, 1, 0x02), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x17), 0xc0}, + {TASDEV_REG_SDW(0, 1, 0x19), 0x60}, + {TASDEV_REG_SDW(0, 1, 0x35), 0x75}, + {TASDEV_REG_SDW(0, 1, 0x3d), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x3e), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x3f), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x40), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x41), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x42), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x43), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x44), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x45), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x47), 0xab}, + {TASDEV_REG_SDW(0, 0xfd, 0x0d), 0x0d}, + {TASDEV_REG_SDW(0, 0xfd, 0x39), 0x00}, + {TASDEV_REG_SDW(0, 0xfd, 0x3e), 0x00}, + {TASDEV_REG_SDW(0, 0xfd, 0x45), 0x00}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS21, 0x02, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS21, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS24, 0x02, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS24, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS26, 0x02, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS26, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS28, 0x02, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS28, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS127, 0x02, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS127, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU21, 0x01, 1), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU21, 0x02, 1), 0x9c00}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x01, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x01, 1), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x0b, 1), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x01, 1), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x01, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x0b, 1), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 1), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 2), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 1), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 2), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x01, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x05, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x01, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x05, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 1), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 2), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 3), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 4), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 5), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 6), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 7), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x06, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT23, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT23, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 1), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 2), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 3), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 4), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 5), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 6), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 7), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 8), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 9), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xa), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xb), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xc), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xd), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xe), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xf), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, 0x1, 0), 0x3}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, 0x10, 0), 0x3}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x06, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x13, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x06, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x13, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x05, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x10, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_TG23, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x01, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x06, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x07, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x09, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x0a, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x10, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x13, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x14, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x15, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x16, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_UDMPU23, 0x10, 0), 0x0}, +}; + +static const struct reg_sequence tas2783_init_seq[] = { + REG_SEQ0(SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x10, 0x00), 0x04), + REG_SEQ0(0x00800418, 0x00), + REG_SEQ0(0x00800419, 0x00), + REG_SEQ0(0x0080041a, 0x00), + REG_SEQ0(0x0080041b, 0x00), + REG_SEQ0(0x00800428, 0x40), + REG_SEQ0(0x00800429, 0x00), + REG_SEQ0(0x0080042a, 0x00), + REG_SEQ0(0x0080042b, 0x00), + REG_SEQ0(SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x1, 0x00), 0x00), + REG_SEQ0(0x0080005c, 0xD9), + REG_SEQ0(0x00800082, 0x20), + REG_SEQ0(0x008000a1, 0x00), + REG_SEQ0(0x00800097, 0xc8), + REG_SEQ0(0x00800099, 0x20), + REG_SEQ0(0x008000c7, 0xaa), + REG_SEQ0(0x008000b5, 0x74), + REG_SEQ0(0x00800082, 0x20), + REG_SEQ0(0x00807e8d, 0x0d), + REG_SEQ0(0x00807eb9, 0x53), + REG_SEQ0(0x00807ebe, 0x42), + REG_SEQ0(0x00807ec5, 0x37), + REG_SEQ0(0x00800066, 0x92), + REG_SEQ0(0x00800003, 0x28), + REG_SEQ0(0x00800004, 0x21), + REG_SEQ0(0x00800005, 0x41), + REG_SEQ0(0x00800006, 0x00), + REG_SEQ0(0x00800007, 0x20), + REG_SEQ0(0x0080000c, 0x10), + REG_SEQ0(0x00800013, 0x08), + REG_SEQ0(0x00800015, 0x00), + REG_SEQ0(0x00800017, 0x80), + REG_SEQ0(0x0080001a, 0x00), + REG_SEQ0(0x0080001b, 0x22), + REG_SEQ0(0x0080001c, 0x36), + REG_SEQ0(0x0080001d, 0x01), + REG_SEQ0(0x0080001f, 0x00), + REG_SEQ0(0x00800020, 0x2e), + REG_SEQ0(0x00800034, 0x06), + REG_SEQ0(0x00800035, 0xb9), + REG_SEQ0(0x00800036, 0xad), + REG_SEQ0(0x00800037, 0xa8), + REG_SEQ0(0x00800038, 0x00), + REG_SEQ0(0x0080003b, 0xfc), + REG_SEQ0(0x0080003d, 0xdd), + REG_SEQ0(0x00800040, 0xf6), + REG_SEQ0(0x00800041, 0x14), + REG_SEQ0(0x0080005c, 0x19), + REG_SEQ0(0x0080005d, 0x80), + REG_SEQ0(0x00800063, 0x48), + REG_SEQ0(0x00800065, 0x08), + REG_SEQ0(0x00800067, 0x00), + REG_SEQ0(0x0080006a, 0x12), + REG_SEQ0(0x0080006b, 0x7b), + REG_SEQ0(0x0080006c, 0x00), + REG_SEQ0(0x0080006d, 0x00), + REG_SEQ0(0x0080006e, 0x1a), + REG_SEQ0(0x0080006f, 0x00), + REG_SEQ0(0x00800070, 0x96), + REG_SEQ0(0x00800071, 0x02), + REG_SEQ0(0x00800073, 0x08), + REG_SEQ0(0x00800075, 0xe0), + REG_SEQ0(0x0080007a, 0x60), + REG_SEQ0(0x008000bd, 0x00), + REG_SEQ0(0x008000be, 0x00), + REG_SEQ0(0x008000bf, 0x00), + REG_SEQ0(0x008000c0, 0x00), + REG_SEQ0(0x008000c1, 0x00), + REG_SEQ0(0x008000c2, 0x00), + REG_SEQ0(0x008000c3, 0x00), + REG_SEQ0(0x008000c4, 0x00), + REG_SEQ0(0x008000c5, 0x00), + REG_SEQ0(0x00800008, 0x49), + REG_SEQ0(0x00800009, 0x02), + REG_SEQ0(0x0080000a, 0x1a), + REG_SEQ0(0x0080000d, 0x93), + REG_SEQ0(0x0080000e, 0x82), + REG_SEQ0(0x0080000f, 0x42), + REG_SEQ0(0x00800010, 0x84), + REG_SEQ0(0x00800014, 0x0a), + REG_SEQ0(0x00800016, 0x00), + REG_SEQ0(0x00800060, 0x21), +}; + +static int tas2783_sdca_mbq_size(struct device *dev, u32 reg) +{ + switch (reg) { + case 0x000 ... 0x080: /* Data port 0. */ + case 0x100 ... 0x140: /* Data port 1. */ + case 0x200 ... 0x240: /* Data port 2. */ + case 0x300 ... 0x340: /* Data port 3. */ + case 0x400 ... 0x440: /* Data port 4. */ + case 0x500 ... 0x540: /* Data port 5. */ + case 0x800000 ... 0x803fff: /* Page 0 ~ 127. */ + case 0x807e80 ... 0x807eff: /* Page 253. */ + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_UDMPU23, + TAS2783_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU21, TAS2783_SDCA_CTL_FU_MUTE, + TAS2783_DEVICE_CHANNEL_LEFT): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, 0x1, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_TG23, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x0a, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x14, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x15, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x16, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT23, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 2): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 3): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 4): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 5): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 6): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 7): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 8): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 9): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xa): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xb): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xc): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xd): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xe): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xf): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS21, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS21, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS24, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS24, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS25, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS25, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS127, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS127, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS26, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS26, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS28, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS28, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x05, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 2): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x01, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x05, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x01, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x04, 0): + return 1; + + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 2): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 3): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 4): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 5): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 6): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 7): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU21, 0x02, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x0b, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 2): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x0b, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x0b, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x07, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x09, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x13, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x13, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x11, 0): + return 2; + + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT23, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x06, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x06, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x13, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x05, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x06, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x06, 0): + return 4; + + default: + return 0; + } +} + +static bool tas2783_readable_register(struct device *dev, unsigned int reg) +{ + return tas2783_sdca_mbq_size(dev, reg) > 0; +} + +static bool tas2783_volatile_register(struct device *dev, u32 reg) +{ + switch (reg) { + case 0x000 ... 0x080: /* Data port 0. */ + case 0x100 ... 0x140: /* Data port 1. */ + case 0x200 ... 0x240: /* Data port 2. */ + case 0x300 ... 0x340: /* Data port 3. */ + case 0x400 ... 0x440: /* Data port 4. */ + case 0x500 ... 0x540: /* Data port 5. */ + case 0x800001: + return true; + + default: + return false; + } +} + +static const struct regmap_config tas_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .reg_defaults = tas2783_reg_default, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_default), + .max_register = 0x41008000 + TASDEV_REG_SDW(0xa1, 0x60, 0x7f), + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct regmap_sdw_mbq_cfg tas2783_mbq_cfg = { + .mbq_size = tas2783_sdca_mbq_size, +}; + +static s32 tas_clamp(s32 val, s32 max, u32 invert) +{ + val = clamp(val, 0, max); + if (invert) + val = max - val; + + return val; +} + +static s32 tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + s32 val, ret; + + ret = regmap_read(tas_dev->regmap, mc->reg, &val); + if (ret < 0) + return ret; + + ucontrol->value.integer.value[0] = + tas_clamp(val, mc->max, mc->invert); + return ret; +} + +static s32 tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + s32 val, ret; + + val = tas_clamp(ucontrol->value.integer.value[0], + mc->max, mc->invert); + ret = regmap_write(tas_dev->regmap, mc->reg, val); + if (ret) + dev_dbg(tas_dev->dev, "put vol %d into %x %x\n", + val, mc->reg, ret); + + return ret; +} + +static s32 tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + u8 mask = 0; + s32 ret, val = 0; + + /* Read current volume from the device. */ + ret = regmap_read(tas_dev->regmap, mc->reg, &val); + if (ret < 0) { + dev_err(tas_dev->dev, "AMP vol read failed, addr=%x err=%d\n", + mc->reg, ret); + return ret; + } + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + ucontrol->value.integer.value[0] = + tas_clamp(val, mc->max, mc->invert); + + return 0; +} + +static s32 tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + u8 mask; + s32 val, ret; + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tas_clamp(ucontrol->value.integer.value[0], mc->max, + mc->invert); + ret = regmap_update_bits(tas_dev->regmap, mc->reg, + mask, val << mc->shift); + if (ret) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + } + + return ret; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, tas2781_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, tas2781_dvc_tlv), +}; + +static s32 tas2783_validate_calibdata(struct tas2783_prv *tas_dev, + u8 *data, u32 size) +{ + u32 ts, spk_count, size_calculated; + u32 crc_calculated, crc_read, i; + u32 *tmp_val; + struct tm tm; + + i = 0; + tmp_val = (u32 *)data; + if (tmp_val[i++] != 2783) { + dev_err(tas_dev->dev, "cal data magic number mismatch"); + return -EINVAL; + } + + spk_count = tmp_val[i++]; + if (spk_count > TAS2783_CALIB_MAX_SPK_COUNT) { + dev_err(tas_dev->dev, "cal data spk_count too large"); + return -EINVAL; + } + + ts = tmp_val[i++]; + time64_to_tm(ts, 0, &tm); + dev_dbg(tas_dev->dev, "cal data timestamp: %ld-%d-%d %d:%d:%d", + tm.tm_year + 1900, tm.tm_mon + 1, tm.tm_mday, + tm.tm_hour, tm.tm_min, tm.tm_sec); + + size_calculated = + (spk_count * TAS2783_CALIB_PARAMS * sizeof(u32)) + + TAS2783_CALIB_HDR_SZ + TAS2783_CALIB_CRC_SZ; + if (size_calculated > TAS2783_CALIB_DATA_SZ) { + dev_err(tas_dev->dev, "cali data sz too large"); + return -EINVAL; + } else if (size < size_calculated) { + dev_err(tas_dev->dev, "cali data size mismatch calc=%u vs %d\n", + size, size_calculated); + return -EINVAL; + } + + crc_calculated = crc32(~0, data, + size_calculated - TAS2783_CALIB_CRC_SZ) ^ ~0; + crc_read = tmp_val[(size_calculated - TAS2783_CALIB_CRC_SZ) / sizeof(u32)]; + if (crc_calculated != crc_read) { + dev_err(tas_dev->dev, + "calib data integrity check fail, 0x%08x vs 0x%08x\n", + crc_calculated, crc_read); + return -EINVAL; + } + + return 0; +} + +static void tas2783_set_calib_params_to_device(struct tas2783_prv *tas_dev, u32 *cali_data) +{ + u32 dev_count, offset, i, device_num; + u32 reg_value; + u8 buf[4]; + + dev_count = cali_data[1]; + offset = 3; + + for (device_num = 0; device_num < dev_count; device_num++) { + if (cali_data[offset] != tas_dev->sdw_peripheral->id.unique_id) { + offset += TAS2783_CALIB_PARAMS; + continue; + } + offset++; + + for (i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + reg_value = cali_data[offset + i]; + buf[0] = reg_value >> 24; + buf[1] = reg_value >> 16; + buf[2] = reg_value >> 8; + buf[3] = reg_value & 0xff; + regmap_bulk_write(tas_dev->regmap, tas2783_cali_reg[i], + buf, sizeof(u32)); + } + break; + } + + if (device_num == dev_count) + dev_err(tas_dev->dev, "device not found\n"); + else + dev_dbg(tas_dev->dev, "calib data update done\n"); +} + +static s32 tas2783_update_calibdata(struct tas2783_prv *tas_dev) +{ + efi_guid_t efi_guid = TAS2783_CALI_GUID; + u32 attr, i, *tmp_val; + unsigned long size; + s32 ret; + efi_status_t status; + static efi_char16_t efi_names[][32] = { + L"SmartAmpCalibrationData", L"CALI_DATA"}; + + tmp_val = (u32 *)tas_dev->cali_data.data; + attr = 0; + i = 0; + + /* + * In some cases, the calibration is performed in Windows, + * and data was saved in UEFI. Linux can access it. + */ + for (i = 0; i < ARRAY_SIZE(efi_names); i++) { + size = 0; + status = efi.get_variable(efi_names[i], &efi_guid, &attr, + &size, NULL); + if (size > TAS2783_CALIB_DATA_SZ) { + dev_err(tas_dev->dev, "cali data too large\n"); + break; + } + + tas_dev->cali_data.read_sz = size; + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_names[i], &efi_guid, &attr, + &tas_dev->cali_data.read_sz, + tas_dev->cali_data.data); + dev_dbg(tas_dev->dev, "cali get %lu bytes result:%ld\n", + tas_dev->cali_data.read_sz, status); + } + if (status == EFI_SUCCESS) + break; + } + + if (status != EFI_SUCCESS) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return 0; + } + + mutex_lock(&tas_dev->calib_lock); + ret = tas2783_validate_calibdata(tas_dev, tas_dev->cali_data.data, + tas_dev->cali_data.read_sz); + if (!ret) + tas2783_set_calib_params_to_device(tas_dev, tmp_val); + mutex_unlock(&tas_dev->calib_lock); + + return ret; +} + +static s32 read_header(const u8 *data, struct bin_header_t *hdr) +{ + hdr->vendor_id = get_unaligned_le16(&data[0]); + hdr->file_id = get_unaligned_le32(&data[2]); + hdr->version = get_unaligned_le16(&data[6]); + hdr->length = get_unaligned_le32(&data[8]); + return 12; +} + +static void tas2783_fw_ready(const struct firmware *fmw, void *context) +{ + struct tas2783_prv *tas_dev = + (struct tas2783_prv *)context; + const u8 *buf = NULL; + s32 offset = 0, img_sz, file_blk_size, ret; + struct bin_header_t hdr; + + if (!fmw || !fmw->data) { + /* No firmware binary, devices will work in ROM mode. */ + dev_err(tas_dev->dev, + "Failed to read %s, no side-effect on driver running\n", + tas_dev->rca_binaryname); + ret = -EINVAL; + goto out; + } + + mutex_lock(&tas_dev->pde_lock); + img_sz = fmw->size; + buf = fmw->data; + offset += FW_DL_OFFSET; + while (offset < (img_sz - FW_FL_HDR)) { + memset(&hdr, 0, sizeof(hdr)); + offset += read_header(&buf[offset], &hdr); + dev_dbg(tas_dev->dev, + "vndr=%d, file=%d, version=%d, len=%d, off=%d\n", + hdr.vendor_id, hdr.file_id, hdr.version, + hdr.length, offset); + /* size also includes the header */ + file_blk_size = hdr.length - FW_FL_HDR; + + switch (hdr.file_id) { + case 0: + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + PRAM_ADDR_START, file_blk_size, + &buf[offset]); + if (ret < 0) + dev_err(tas_dev->dev, + "PRAM update failed: %d", ret); + break; + + case 1: + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + YRAM_ADDR_START, file_blk_size, + &buf[offset]); + if (ret < 0) + dev_err(tas_dev->dev, + "YRAM update failed: %d", ret); + + break; + + default: + ret = -EINVAL; + dev_err(tas_dev->dev, "Unsupported file"); + break; + } + + if (ret == 0) + offset += file_blk_size; + else + break; + }; + mutex_unlock(&tas_dev->pde_lock); + tas2783_update_calibdata(tas_dev); + +out: + if (!ret) + tas_dev->fw_dl_success = true; + tas_dev->fw_dl_task_done = true; + wake_up(&tas_dev->fw_wait); + if (fmw) + release_firmware(fmw); +} + +static inline s32 tas_clear_latch(struct tas2783_prv *priv) +{ + return regmap_update_bits(priv->regmap, + TASDEV_REG_SDW(0, 0, 0x5c), + 0x04, 0x04); +} + +static s32 tas_fu21_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, s32 event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct tas2783_prv *tas_dev = snd_soc_component_get_drvdata(component); + s32 mute; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + mute = 1; + break; + } + + return sdw_write_no_pm(tas_dev->sdw_peripheral, + SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 1), mute); +} + +static s32 tas_fu23_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, s32 event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct tas2783_prv *tas_dev = snd_soc_component_get_drvdata(component); + s32 mute; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + mute = 1; + break; + } + + return sdw_write_no_pm(tas_dev->sdw_peripheral, + SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 1), mute); +} + +static const struct snd_soc_dapm_widget tas_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_DAC_E("FU21", NULL, SND_SOC_NOPM, 0, 0, tas_fu21_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("FU23", NULL, SND_SOC_NOPM, 0, 0, tas_fu23_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tas_audio_map[] = { + {"FU21", NULL, "ASI"}, + {"SPK", NULL, "FU21"}, + {"FU23", NULL, "ASI"}, + {"SPK", NULL, "FU23"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static s32 tas_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, s32 direction) +{ + if (!sdw_stream) + return 0; + + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tas_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static s32 tas_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + struct sdw_slave *sdw_peripheral = tas_dev->sdw_peripheral; + s32 ret, retry = 3; + + if (!tas_dev->fw_dl_success) { + dev_err(tas_dev->dev, "error playback without fw download"); + return -EINVAL; + } + + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + if (!sdw_stream) + return -EINVAL; + + ret = tas_clear_latch(tas_dev); + if (ret) + dev_err(tas_dev->dev, + "clear latch failed, err=%d", ret); + + mutex_lock(&tas_dev->pde_lock); + /* + * Sometimes, there is error returned during power on. + * So added retry logic to ensure power on so that + * port prepare succeeds + */ + do { + ret = regmap_write(tas_dev->regmap, + SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POW_STATE, 0), + TAS2783_SDCA_POW_STATE_ON); + if (!ret) + break; + usleep_range(2000, 2200); + } while (retry--); + mutex_unlock(&tas_dev->pde_lock); + if (ret) + return ret; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) + dev_err(dai->dev, "Unable to configure port\n"); + + return ret; +} + +static s32 tas_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + s32 ret; + struct snd_soc_component *component = dai->component; + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + sdw_stream_remove_slave(tas_dev->sdw_peripheral, sdw_stream); + + mutex_lock(&tas_dev->pde_lock); + ret = regmap_write(tas_dev->regmap, + SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POW_STATE, 0), + TAS2783_SDCA_POW_STATE_OFF); + mutex_unlock(&tas_dev->pde_lock); + + return ret; +} + +static const struct snd_soc_dai_ops tas_dai_ops = { + .hw_params = tas_sdw_hw_params, + .hw_free = tas_sdw_pcm_hw_free, + .set_stream = tas_set_sdw_stream, + .shutdown = tas_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tas_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tas_dai_ops, + .symmetric_rate = 1, + }, +}; + +static s32 tas_component_probe(struct snd_soc_component *component) +{ + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + + tas_dev->component = component; + tas25xx_register_misc(tas_dev->sdw_peripheral); + + return 0; +} + +static void tas_component_remove(struct snd_soc_component *codec) +{ + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(codec); + tas25xx_deregister_misc(); + tas_dev->component = NULL; +} + +static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { + .probe = tas_component_probe, + .remove = tas_component_remove, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tas_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas_dapm_widgets), + .dapm_routes = tas_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static s32 tas_init(struct tas2783_prv *tas_dev) +{ + s32 ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tas_dai_driver, + ARRAY_SIZE(tas_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return ret; +} + +static s32 tas_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + s32 nval; + s32 i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = false; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = false; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 200; + + return 0; +} + +static s32 tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tas2783_prv *tas_dev = dev_get_drvdata(dev); + + if (!tas_dev->hw_init) + return 0; + + regcache_cache_only(tas_dev->regmap, true); + return 0; +} + +static s32 tas2783_sdca_dev_system_suspend(struct device *dev) +{ + return tas2783_sdca_dev_suspend(dev); +} + +static s32 tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tas2783_prv *tas_dev = dev_get_drvdata(dev); + unsigned long t; + + if (!slave->unattach_request) + goto regmap_sync; + + t = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT)); + if (!t) { + dev_err(&slave->dev, "resume: initialization timed out\n"); + sdw_show_ping_status(slave->bus, true); + return -ETIMEDOUT; + } + + slave->unattach_request = 0; + +regmap_sync: + regcache_cache_only(tas_dev->regmap, false); + regcache_sync(tas_dev->regmap); + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_system_suspend, tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, tas2783_sdca_dev_resume, NULL) +}; + +static s32 tas_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tas2783_prv *tas_dev = dev_get_drvdata(dev); + s32 ret; + u8 unique_id = tas_dev->sdw_peripheral->id.unique_id; + + if (tas_dev->hw_init) + return 0; + + tas_dev->fw_dl_task_done = false; + tas_dev->fw_dl_success = false; + scnprintf(tas_dev->rca_binaryname, sizeof(tas_dev->rca_binaryname), + "tas2783-%01x.bin", unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, + GFP_KERNEL, tas_dev, tas2783_fw_ready); + if (ret) { + dev_err(tas_dev->dev, + "firmware request failed for uid=%d, ret=%d\n", + unique_id, ret); + return ret; + } + + ret = wait_event_timeout(tas_dev->fw_wait, tas_dev->fw_dl_task_done, + msecs_to_jiffies(TIMEOUT_FW_DL_MS)); + if (!ret) { + dev_err(tas_dev->dev, "fw request, wait_event timeout\n"); + ret = -EAGAIN; + } else { + ret = regmap_multi_reg_write(tas_dev->regmap, tas2783_init_seq, + ARRAY_SIZE(tas2783_init_seq)); + tas_dev->hw_init = true; + } + + return ret; +} + +static s32 tas_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tas2783_prv *tas_dev = dev_get_drvdata(&slave->dev); + struct device *dev = &slave->dev; + + dev_dbg(dev, "Perifpheral status = %s", + status == SDW_SLAVE_UNATTACHED ? "unattached" : + status == SDW_SLAVE_ATTACHED ? "attached" : "alert"); + + tas_dev->status = status; + if (status == SDW_SLAVE_UNATTACHED) + tas_dev->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_dev->hw_init || tas_dev->status != SDW_SLAVE_ATTACHED) + return 0; + + /* updated the cache data to device */ + regcache_cache_only(tas_dev->regmap, false); + regcache_sync(tas_dev->regmap); + + /* perform I/O transfers required for Slave initialization */ + return tas_io_init(&slave->dev, slave); +} + +static const struct sdw_slave_ops tas_sdw_ops = { + .read_prop = tas_read_prop, + .update_status = tas_update_status, +}; + +static void tas_remove(struct tas2783_prv *tas_dev) +{ + snd_soc_unregister_component(tas_dev->dev); +} + +static s32 tas_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + struct device *dev = &peripheral->dev; + struct tas2783_prv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return dev_err_probe(dev, -ENOMEM, + "Failed devm_kzalloc"); + + tas_dev->dev = dev; + tas_dev->sdw_peripheral = peripheral; + tas_dev->hw_init = false; + mutex_init(&tas_dev->calib_lock); + mutex_init(&tas_dev->pde_lock); + + init_waitqueue_head(&tas_dev->fw_wait); + dev_set_drvdata(dev, tas_dev); + regmap = devm_regmap_init_sdw_mbq_cfg(peripheral, + &tas_regmap, + &tas2783_mbq_cfg); + if (IS_ERR(regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + /* keep in cache until the device is fully initialized */ + regcache_cache_only(regmap, true); + tas_dev->regmap = regmap; + return tas_init(tas_dev); +} + +static s32 tas_sdw_remove(struct sdw_slave *peripheral) +{ + struct tas2783_prv *tas_dev = dev_get_drvdata(&peripheral->dev); + + pm_runtime_disable(tas_dev->dev); + tas_remove(tas_dev); + mutex_destroy(&tas_dev->calib_lock); + mutex_destroy(&tas_dev->pde_lock); + dev_set_drvdata(&peripheral->dev, NULL); + + return 0; +} + +static const struct sdw_device_id tas_sdw_id[] = { + /* chipid for the TAS2783 is 0x0000 */ + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, tas_sdw_id); + +static struct sdw_driver tas_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = pm_ptr(&tas2783_sdca_pm), + }, + .probe = tas_sdw_probe, + .remove = tas_sdw_remove, + .ops = &tas_sdw_ops, + .id_table = tas_sdw_id, +}; +module_sdw_driver(tas_sdw_driver); + +MODULE_AUTHOR("Texas Instruments Inc."); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000..a77e332de --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,107 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2025 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2783 driver implements a flexible and configurable + * algo coefficient setting for single TAS2783 chips. + * + * Author: Niranjan H Y <niranjanhy(a)ti.com> + * Author: Baojun Xu <baojun.xu(a)ti.com> + */ +#include <linux/workqueue.h> + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* book, page, register */ +#define TASDEV_REG_SDW(book, page, reg) (((book) * 256 * 128) + \ + 0x800000 + ((page) * 128) + (reg)) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEV_REG_SDW(0x0, 0x00, 0x1A) +#define TAS2783_AMP_LEVEL TASDEV_REG_SDW(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +#define PRAM_ADDR_START TASDEV_REG_SDW(0x8c, 0x01, 0x8) +#define PRAM_ADDR_END TASDEV_REG_SDW(0x8c, 0xff, 0x7f) +#define YRAM_ADDR_START TASDEV_REG_SDW(0x00, 0x02, 0x8) +#define YRAM_ADDR_END TASDEV_REG_SDW(0x00, 0x37, 0x7f) + +/* Calibration data */ +#define TAS2783_CAL_R0 TASDEV_REG_SDW(0, 0x16, 0x4C) +#define TAS2783_CAL_INVR0 TASDEV_REG_SDW(0, 0x16, 0x5C) +#define TAS2783_CAL_R0LOW TASDEV_REG_SDW(0, 0x16, 0x64) +#define TAS2783_CAL_POWER TASDEV_REG_SDW(0, 0x15, 0x44) +#define TAS2783_CAL_TLIM TASDEV_REG_SDW(0, 0x17, 0x58) + +/* TAS2783 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ + +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 +#define TAS2783_SDCA_ENT_FU26 0x03 +#define TAS2783_SDCA_ENT_XU22 0x04 +#define TAS2783_SDCA_ENT_CS24 0x05 +#define TAS2783_SDCA_ENT_CS21 0x06 +#define TAS2783_SDCA_ENT_CS25 0x07 +#define TAS2783_SDCA_ENT_CS26 0x08 +#define TAS2783_SDCA_ENT_CS28 0x09 +#define TAS2783_SDCA_ENT_PDE23 0x0C +#define TAS2783_SDCA_ENT_UDMPU23 0x0E +#define TAS2783_SDCA_ENT_SAPU29 0x0F +#define TAS2783_SDCA_ENT_PPU21 0x10 +#define TAS2783_SDCA_ENT_PPU26 0x11 +#define TAS2783_SDCA_ENT_TG23 0x12 +#define TAS2783_SDCA_ENT_IT21 0x13 +#define TAS2783_SDCA_ENT_IT29 0x14 +#define TAS2783_SDCA_ENT_IT26 0x15 +#define TAS2783_SDCA_ENT_IT28 0x16 +#define TAS2783_SDCA_ENT_OT24 0x17 +#define TAS2783_SDCA_ENT_OT23 0x18 +#define TAS2783_SDCA_ENT_OT25 0x19 +#define TAS2783_SDCA_ENT_OT28 0x1A +#define TAS2783_SDCA_ENT_MU26 0x1b +#define TAS2783_SDCA_ENT_OT127 0x1E +#define TAS2783_SDCA_ENT_FU127 0x1F +#define TAS2783_SDCA_ENT_CS127 0x20 +#define TAS2783_SDCA_ENT_MFPU21 0x22 +#define TAS2783_SDCA_ENT_MFPU26 0x23 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POW_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 + +#define TAS2783_SDCA_POW_STATE_ON 0 +#define TAS2783_SDCA_POW_STATE_OFF 3 + +/* calibration data */ +#define TAS2783_CALIB_PARAMS 6 /* 5 + 1 unique id */ +#define TAS2783_CALIB_MAX_SPK_COUNT 8 +#define TAS2783_CALIB_HDR_SZ 12 +#define TAS2783_CALIB_CRC_SZ 4 +#define TAS2783_CALIB_DATA_SZ ((TAS2783_CALIB_HDR_SZ) + TAS2783_CALIB_CRC_SZ + \ + ((TAS2783_CALIB_PARAMS) * 4 * (TAS2783_CALIB_MAX_SPK_COUNT))) + +#if IS_ENABLED(CONFIG_SND_SOC_TAS2783_UTIL) +int32_t tas25xx_register_misc(struct sdw_slave *peripheral); +int32_t tas25xx_deregister_misc(void); +#endif + +#endif /*__TAS2783_H__ */ -- 2.45.2

2 4

[PATCH v2 1/4] ASoc: tas2783A: Add soundwire based codec driver
by Niranjan H Y 10 Sep '25

10 Sep '25

TAS2783 is mono digital input class-D Smart Amplifier based on MIPI Alliance Soundwire interface. The driver supports loading the algorithm coefficients for one or more tas2783 chips. Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v2: - append Volume to amp gain mixer control - remove unwanted debug log --- sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 1417 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 107 +++ 4 files changed, 1540 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b8d58d2fe326..1f012e6a6c9e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -266,6 +266,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2770 imply SND_SOC_TAS2780 imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783_SDW imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -2074,6 +2075,19 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783_SDW + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + select SOUNDWIRE + select REGMAP + select REGMAP_SOUNDWIRE + select EFI + select CRC32 + help + Enable support for Texas Instruments TAS2783A Digital input + mono Class-D and DSP-inside audio power amplifiers. TAS2783 + driver implements a flexible and configurable algorithm + cofficient setting, for one, two or multiple TAS2783 chips. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a476d6c45451..1920fa17be8d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -316,6 +316,7 @@ snd-soc-tas2781-comlib-y := tas2781-comlib.o snd-soc-tas2781-comlib-i2c-y := tas2781-comlib-i2c.o snd-soc-tas2781-fmwlib-y := tas2781-fmwlib.o snd-soc-tas2781-i2c-y := tas2781-i2c.o +snd-soc-tas2783-sdw-y := tas2783-sdw.o snd-soc-tfa9879-y := tfa9879.o snd-soc-tfa989x-y := tfa989x.o snd-soc-tlv320adc3xxx-y := tlv320adc3xxx.o @@ -732,6 +733,7 @@ obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB_I2C) += snd-soc-tas2781-comlib-i2c.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783_SDW) += snd-soc-tas2783-sdw.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..b5abbb1ddee4 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,1417 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2783 driver implements a flexible and configurable +// algo coefficient setting for single TAS2783 chips. +// +// Author: Niranjan H Y <niranjanhy(a)ti.com> +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> + +#include <linux/unaligned.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/wait.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +#define TIMEOUT_FW_DL_MS (3000) +#define FW_DL_OFFSET 36 +#define FW_FL_HDR 12 +#define TAS2783_PROBE_TIMEOUT 5000 +#define TAS2783_CALI_GUID EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, \ + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92) + +#if !IS_ENABLED(CONFIG_SND_SOC_TAS2783_UTIL) +static void tas25xx_register_misc(struct sdw_slave *peripheral) {} +static void tas25xx_deregister_misc(void) {} +#endif + +static const u32 tas2783_cali_reg[] = { + TAS2783_CAL_R0, + TAS2783_CAL_INVR0, + TAS2783_CAL_R0LOW, + TAS2783_CAL_POWER, + TAS2783_CAL_TLIM, +}; + +struct bin_header_t { + u16 vendor_id; + u16 version; + u32 file_id; + u32 length; +}; + +struct calibration_data { + u32 is_valid; + unsigned long read_sz; + u8 data[TAS2783_CALIB_DATA_SZ]; +}; + +struct tas2783_prv { + struct snd_soc_component *component; + struct calibration_data cali_data; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + /* calibration */ + struct mutex calib_lock; + /* pde and firmware download */ + struct mutex pde_lock; + struct regmap *regmap; + struct device *dev; + struct class *class; + struct attribute_group *cal_attr_groups; + struct tm tm; + u8 rca_binaryname[64]; + u8 dev_name[32]; + bool hw_init; + /* wq for firmware download */ + wait_queue_head_t fw_wait; + bool fw_dl_task_done; + bool fw_dl_success; +}; + +static const struct reg_default tas2783_reg_default[] = { + {TAS2783_AMP_LEVEL, 0x28}, + {TASDEV_REG_SDW(0, 0, 0x03), 0x28}, + {TASDEV_REG_SDW(0, 0, 0x04), 0x21}, + {TASDEV_REG_SDW(0, 0, 0x05), 0x41}, + {TASDEV_REG_SDW(0, 0, 0x06), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x07), 0x20}, + {TASDEV_REG_SDW(0, 0, 0x08), 0x09}, + {TASDEV_REG_SDW(0, 0, 0x09), 0x02}, + {TASDEV_REG_SDW(0, 0, 0x0a), 0x0a}, + {TASDEV_REG_SDW(0, 0, 0x0c), 0x10}, + {TASDEV_REG_SDW(0, 0, 0x0d), 0x13}, + {TASDEV_REG_SDW(0, 0, 0x0e), 0xc2}, + {TASDEV_REG_SDW(0, 0, 0x0f), 0x40}, + {TASDEV_REG_SDW(0, 0, 0x10), 0x04}, + {TASDEV_REG_SDW(0, 0, 0x13), 0x13}, + {TASDEV_REG_SDW(0, 0, 0x14), 0x12}, + {TASDEV_REG_SDW(0, 0, 0x15), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x16), 0x12}, + {TASDEV_REG_SDW(0, 0, 0x17), 0x80}, + {TAS2783_DVC_LVL, 0x00}, + {TASDEV_REG_SDW(0, 0, 0x1b), 0x61}, + {TASDEV_REG_SDW(0, 0, 0x1c), 0x36}, + {TASDEV_REG_SDW(0, 0, 0x1d), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x1f), 0x01}, + {TASDEV_REG_SDW(0, 0, 0x20), 0x2e}, + {TASDEV_REG_SDW(0, 0, 0x21), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x34), 0x06}, + {TASDEV_REG_SDW(0, 0, 0x35), 0xbd}, + {TASDEV_REG_SDW(0, 0, 0x36), 0xad}, + {TASDEV_REG_SDW(0, 0, 0x37), 0xa8}, + {TASDEV_REG_SDW(0, 0, 0x38), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x3b), 0xfc}, + {TASDEV_REG_SDW(0, 0, 0x3d), 0xdd}, + {TASDEV_REG_SDW(0, 0, 0x40), 0xf6}, + {TASDEV_REG_SDW(0, 0, 0x41), 0x14}, + {TASDEV_REG_SDW(0, 0, 0x5c), 0x19}, + {TASDEV_REG_SDW(0, 0, 0x5d), 0x80}, + {TASDEV_REG_SDW(0, 0, 0x63), 0x48}, + {TASDEV_REG_SDW(0, 0, 0x65), 0x08}, + {TASDEV_REG_SDW(0, 0, 0x66), 0xb2}, + {TASDEV_REG_SDW(0, 0, 0x67), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x6a), 0x12}, + {TASDEV_REG_SDW(0, 0, 0x6b), 0xfb}, + {TASDEV_REG_SDW(0, 0, 0x6c), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x6d), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x6e), 0x1a}, + {TASDEV_REG_SDW(0, 0, 0x6f), 0x00}, + {TASDEV_REG_SDW(0, 0, 0x70), 0x96}, + {TASDEV_REG_SDW(0, 0, 0x71), 0x02}, + {TASDEV_REG_SDW(0, 0, 0x73), 0x08}, + {TASDEV_REG_SDW(0, 0, 0x75), 0xe0}, + {TASDEV_REG_SDW(0, 0, 0x7a), 0x60}, + {TASDEV_REG_SDW(0, 0, 0x60), 0x21}, + {TASDEV_REG_SDW(0, 1, 0x02), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x17), 0xc0}, + {TASDEV_REG_SDW(0, 1, 0x19), 0x60}, + {TASDEV_REG_SDW(0, 1, 0x35), 0x75}, + {TASDEV_REG_SDW(0, 1, 0x3d), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x3e), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x3f), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x40), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x41), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x42), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x43), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x44), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x45), 0x00}, + {TASDEV_REG_SDW(0, 1, 0x47), 0xab}, + {TASDEV_REG_SDW(0, 0xfd, 0x0d), 0x0d}, + {TASDEV_REG_SDW(0, 0xfd, 0x39), 0x00}, + {TASDEV_REG_SDW(0, 0xfd, 0x3e), 0x00}, + {TASDEV_REG_SDW(0, 0xfd, 0x45), 0x00}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS21, 0x02, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS21, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS24, 0x02, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS24, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS26, 0x02, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS26, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS28, 0x02, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS28, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS127, 0x02, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS127, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU21, 0x01, 1), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU21, 0x02, 1), 0x9c00}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x01, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x01, 1), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x0b, 1), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x01, 1), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x01, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x0b, 1), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 1), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 2), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 1), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 2), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x01, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x05, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x01, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x05, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 1), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 2), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 3), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 4), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 5), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 6), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 7), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x06, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT23, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT23, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x04, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 1), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 2), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 3), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 4), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 5), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 6), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 7), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 8), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 9), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xa), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xb), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xc), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xd), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xe), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xf), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, 0x1, 0), 0x3}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, 0x10, 0), 0x3}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x06, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x13, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x06, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x13, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x05, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x10, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x11, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_TG23, 0x10, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x01, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x06, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x07, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x08, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x09, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x0a, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x10, 0), 0x1}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x12, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x13, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x14, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x15, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x16, 0), 0x0}, + {SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_UDMPU23, 0x10, 0), 0x0}, +}; + +static const struct reg_sequence tas2783_init_seq[] = { + REG_SEQ0(SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x10, 0x00), 0x04), + REG_SEQ0(0x00800418, 0x00), + REG_SEQ0(0x00800419, 0x00), + REG_SEQ0(0x0080041a, 0x00), + REG_SEQ0(0x0080041b, 0x00), + REG_SEQ0(0x00800428, 0x40), + REG_SEQ0(0x00800429, 0x00), + REG_SEQ0(0x0080042a, 0x00), + REG_SEQ0(0x0080042b, 0x00), + REG_SEQ0(SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x1, 0x00), 0x00), + REG_SEQ0(0x0080005c, 0xD9), + REG_SEQ0(0x00800082, 0x20), + REG_SEQ0(0x008000a1, 0x00), + REG_SEQ0(0x00800097, 0xc8), + REG_SEQ0(0x00800099, 0x20), + REG_SEQ0(0x008000c7, 0xaa), + REG_SEQ0(0x008000b5, 0x74), + REG_SEQ0(0x00800082, 0x20), + REG_SEQ0(0x00807e8d, 0x0d), + REG_SEQ0(0x00807eb9, 0x53), + REG_SEQ0(0x00807ebe, 0x42), + REG_SEQ0(0x00807ec5, 0x37), + REG_SEQ0(0x00800066, 0x92), + REG_SEQ0(0x00800003, 0x28), + REG_SEQ0(0x00800004, 0x21), + REG_SEQ0(0x00800005, 0x41), + REG_SEQ0(0x00800006, 0x00), + REG_SEQ0(0x00800007, 0x20), + REG_SEQ0(0x0080000c, 0x10), + REG_SEQ0(0x00800013, 0x08), + REG_SEQ0(0x00800015, 0x00), + REG_SEQ0(0x00800017, 0x80), + REG_SEQ0(0x0080001a, 0x00), + REG_SEQ0(0x0080001b, 0x22), + REG_SEQ0(0x0080001c, 0x36), + REG_SEQ0(0x0080001d, 0x01), + REG_SEQ0(0x0080001f, 0x00), + REG_SEQ0(0x00800020, 0x2e), + REG_SEQ0(0x00800034, 0x06), + REG_SEQ0(0x00800035, 0xb9), + REG_SEQ0(0x00800036, 0xad), + REG_SEQ0(0x00800037, 0xa8), + REG_SEQ0(0x00800038, 0x00), + REG_SEQ0(0x0080003b, 0xfc), + REG_SEQ0(0x0080003d, 0xdd), + REG_SEQ0(0x00800040, 0xf6), + REG_SEQ0(0x00800041, 0x14), + REG_SEQ0(0x0080005c, 0x19), + REG_SEQ0(0x0080005d, 0x80), + REG_SEQ0(0x00800063, 0x48), + REG_SEQ0(0x00800065, 0x08), + REG_SEQ0(0x00800067, 0x00), + REG_SEQ0(0x0080006a, 0x12), + REG_SEQ0(0x0080006b, 0x7b), + REG_SEQ0(0x0080006c, 0x00), + REG_SEQ0(0x0080006d, 0x00), + REG_SEQ0(0x0080006e, 0x1a), + REG_SEQ0(0x0080006f, 0x00), + REG_SEQ0(0x00800070, 0x96), + REG_SEQ0(0x00800071, 0x02), + REG_SEQ0(0x00800073, 0x08), + REG_SEQ0(0x00800075, 0xe0), + REG_SEQ0(0x0080007a, 0x60), + REG_SEQ0(0x008000bd, 0x00), + REG_SEQ0(0x008000be, 0x00), + REG_SEQ0(0x008000bf, 0x00), + REG_SEQ0(0x008000c0, 0x00), + REG_SEQ0(0x008000c1, 0x00), + REG_SEQ0(0x008000c2, 0x00), + REG_SEQ0(0x008000c3, 0x00), + REG_SEQ0(0x008000c4, 0x00), + REG_SEQ0(0x008000c5, 0x00), + REG_SEQ0(0x00800008, 0x49), + REG_SEQ0(0x00800009, 0x02), + REG_SEQ0(0x0080000a, 0x1a), + REG_SEQ0(0x0080000d, 0x93), + REG_SEQ0(0x0080000e, 0x82), + REG_SEQ0(0x0080000f, 0x42), + REG_SEQ0(0x00800010, 0x84), + REG_SEQ0(0x00800014, 0x0a), + REG_SEQ0(0x00800016, 0x00), + REG_SEQ0(0x00800060, 0x21), +}; + +static int tas2783_sdca_mbq_size(struct device *dev, u32 reg) +{ + switch (reg) { + case 0x000 ... 0x080: /* Data port 0. */ + case 0x100 ... 0x140: /* Data port 1. */ + case 0x200 ... 0x240: /* Data port 2. */ + case 0x300 ... 0x340: /* Data port 3. */ + case 0x400 ... 0x440: /* Data port 4. */ + case 0x500 ... 0x540: /* Data port 5. */ + case 0x800000 ... 0x803fff: /* Page 0 ~ 127. */ + case 0x807e80 ... 0x807eff: /* Page 253. */ + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_UDMPU23, + TAS2783_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU21, TAS2783_SDCA_CTL_FU_MUTE, + TAS2783_DEVICE_CHANNEL_LEFT): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, 0x1, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_TG23, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x0a, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x14, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x15, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x16, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT23, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 2): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 3): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 4): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 5): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 6): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 7): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 8): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 9): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xa): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xb): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xc): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xd): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xe): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x12, 0xf): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS21, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS21, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS24, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS24, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS25, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS25, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS127, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS127, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS26, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS26, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS28, 0x02, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_CS28, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x05, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 2): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x01, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x04, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x05, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x01, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x04, 0): + return 1; + + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x11, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 2): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 3): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 4): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 5): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 6): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x01, 7): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU21, 0x02, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x0b, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 2): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x0b, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x0b, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x0b, 1): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x07, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x09, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x13, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x13, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x11, 0): + return 2; + + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT21, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT26, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT28, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_IT29, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT23, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT24, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT25, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT28, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_OT127, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MU26, 0x06, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU127, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU26, 0x10, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x06, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x12, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_XU22, 0x13, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU21, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_MFPU26, 0x08, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_SAPU29, 0x05, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU21, 0x06, 0): + case SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PPU26, 0x06, 0): + return 4; + + default: + return 0; + } +} + +static bool tas2783_readable_register(struct device *dev, unsigned int reg) +{ + return tas2783_sdca_mbq_size(dev, reg) > 0; +} + +static bool tas2783_volatile_register(struct device *dev, u32 reg) +{ + switch (reg) { + case 0x000 ... 0x080: /* Data port 0. */ + case 0x100 ... 0x140: /* Data port 1. */ + case 0x200 ... 0x240: /* Data port 2. */ + case 0x300 ... 0x340: /* Data port 3. */ + case 0x400 ... 0x440: /* Data port 4. */ + case 0x500 ... 0x540: /* Data port 5. */ + case 0x800001: + return true; + + default: + return false; + } +} + +static const struct regmap_config tas_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .reg_defaults = tas2783_reg_default, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_default), + .max_register = 0x41008000 + TASDEV_REG_SDW(0xa1, 0x60, 0x7f), + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct regmap_sdw_mbq_cfg tas2783_mbq_cfg = { + .mbq_size = tas2783_sdca_mbq_size, +}; + +static s32 tas_clamp(s32 val, s32 max, u32 invert) +{ + val = clamp(val, 0, max); + if (invert) + val = max - val; + + return val; +} + +static s32 tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + s32 val, ret; + + ret = regmap_read(tas_dev->regmap, mc->reg, &val); + if (ret < 0) + return ret; + + ucontrol->value.integer.value[0] = + tas_clamp(val, mc->max, mc->invert); + return ret; +} + +static s32 tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + s32 val, ret; + + val = tas_clamp(ucontrol->value.integer.value[0], + mc->max, mc->invert); + ret = regmap_write(tas_dev->regmap, mc->reg, val); + if (ret) + dev_dbg(tas_dev->dev, "put vol %d into %x %x\n", + val, mc->reg, ret); + + return ret; +} + +static s32 tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + u8 mask = 0; + s32 ret, val = 0; + + /* Read current volume from the device. */ + ret = regmap_read(tas_dev->regmap, mc->reg, &val); + if (ret < 0) { + dev_err(tas_dev->dev, "AMP vol read failed, addr=%x err=%d\n", + mc->reg, ret); + return ret; + } + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + ucontrol->value.integer.value[0] = + tas_clamp(val, mc->max, mc->invert); + + return 0; +} + +static s32 tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + u8 mask; + s32 val, ret; + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tas_clamp(ucontrol->value.integer.value[0], mc->max, + mc->invert); + ret = regmap_update_bits(tas_dev->regmap, mc->reg, + mask, val << mc->shift); + if (ret) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + } + + return ret; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, tas2781_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, tas2781_dvc_tlv), +}; + +static s32 tas2783_validate_calibdata(struct tas2783_prv *tas_dev, + u8 *data, u32 size) +{ + u32 ts, spk_count, size_calculated; + u32 crc_calculated, crc_read, i; + u32 *tmp_val; + struct tm tm; + + i = 0; + tmp_val = (u32 *)data; + if (tmp_val[i++] != 2783) { + dev_err(tas_dev->dev, "cal data magic number mismatch"); + return -EINVAL; + } + + spk_count = tmp_val[i++]; + if (spk_count > TAS2783_CALIB_MAX_SPK_COUNT) { + dev_err(tas_dev->dev, "cal data spk_count too large"); + return -EINVAL; + } + + ts = tmp_val[i++]; + time64_to_tm(ts, 0, &tm); + dev_dbg(tas_dev->dev, "cal data timestamp: %ld-%d-%d %d:%d:%d", + tm.tm_year + 1900, tm.tm_mon + 1, tm.tm_mday, + tm.tm_hour, tm.tm_min, tm.tm_sec); + + size_calculated = + (spk_count * TAS2783_CALIB_PARAMS * sizeof(u32)) + + TAS2783_CALIB_HDR_SZ + TAS2783_CALIB_CRC_SZ; + if (size_calculated > TAS2783_CALIB_DATA_SZ) { + dev_err(tas_dev->dev, "cali data sz too large"); + return -EINVAL; + } else if (size < size_calculated) { + dev_err(tas_dev->dev, "cali data size mismatch calc=%u vs %d\n", + size, size_calculated); + return -EINVAL; + } + + crc_calculated = crc32(~0, data, + size_calculated - TAS2783_CALIB_CRC_SZ) ^ ~0; + crc_read = tmp_val[(size_calculated - TAS2783_CALIB_CRC_SZ) / sizeof(u32)]; + if (crc_calculated != crc_read) { + dev_err(tas_dev->dev, + "calib data integrity check fail, 0x%08x vs 0x%08x\n", + crc_calculated, crc_read); + return -EINVAL; + } + + return 0; +} + +static void tas2783_set_calib_params_to_device(struct tas2783_prv *tas_dev, u32 *cali_data) +{ + u32 dev_count, offset, i, device_num; + u32 reg_value; + u8 buf[4]; + + dev_count = cali_data[1]; + offset = 3; + + for (device_num = 0; device_num < dev_count; device_num++) { + if (cali_data[offset] != tas_dev->sdw_peripheral->id.unique_id) { + offset += TAS2783_CALIB_PARAMS; + continue; + } + offset++; + + for (i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + reg_value = cali_data[offset + i]; + buf[0] = reg_value >> 24; + buf[1] = reg_value >> 16; + buf[2] = reg_value >> 8; + buf[3] = reg_value & 0xff; + regmap_bulk_write(tas_dev->regmap, tas2783_cali_reg[i], + buf, sizeof(u32)); + } + break; + } + + if (device_num == dev_count) + dev_err(tas_dev->dev, "device not found\n"); + else + dev_dbg(tas_dev->dev, "calib data update done\n"); +} + +static s32 tas2783_update_calibdata(struct tas2783_prv *tas_dev) +{ + efi_guid_t efi_guid = TAS2783_CALI_GUID; + u32 attr, i, *tmp_val; + size_t size; + s32 ret; + efi_status_t status; + static efi_char16_t efi_names[][32] = { + L"SmartAmpCalibrationData", L"CALI_DATA"}; + + tmp_val = (u32 *)tas_dev->cali_data.data; + attr = 0; + i = 0; + + /* + * In some cases, the calibration is performed in Windows, + * and data was saved in UEFI. Linux can access it. + */ + for (i = 0; i < ARRAY_SIZE(efi_names); i++) { + size = 0; + status = efi.get_variable(efi_names[i], &efi_guid, &attr, + &size, NULL); + if (size > TAS2783_CALIB_DATA_SZ) { + dev_err(tas_dev->dev, "cali data too large\n"); + break; + } + + tas_dev->cali_data.read_sz = size; + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_names[i], &efi_guid, &attr, + &tas_dev->cali_data.read_sz, + tas_dev->cali_data.data); + dev_dbg(tas_dev->dev, "cali get %lu bytes result:%ld\n", + tas_dev->cali_data.read_sz, status); + } + if (status == EFI_SUCCESS) + break; + } + + if (status != EFI_SUCCESS) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return 0; + } + + mutex_lock(&tas_dev->calib_lock); + ret = tas2783_validate_calibdata(tas_dev, tas_dev->cali_data.data, + tas_dev->cali_data.read_sz); + if (!ret) + tas2783_set_calib_params_to_device(tas_dev, tmp_val); + mutex_unlock(&tas_dev->calib_lock); + + return ret; +} + +static s32 read_header(const u8 *data, struct bin_header_t *hdr) +{ + hdr->vendor_id = get_unaligned_le16(&data[0]); + hdr->file_id = get_unaligned_le32(&data[2]); + hdr->version = get_unaligned_le16(&data[6]); + hdr->length = get_unaligned_le32(&data[8]); + return 12; +} + +static void tas2783_fw_ready(const struct firmware *fmw, void *context) +{ + struct tas2783_prv *tas_dev = + (struct tas2783_prv *)context; + const u8 *buf = NULL; + s32 offset = 0, img_sz, file_blk_size, ret; + struct bin_header_t hdr; + + if (!fmw || !fmw->data) { + /* No firmware binary, devices will work in ROM mode. */ + dev_err(tas_dev->dev, + "Failed to read %s, no side-effect on driver running\n", + tas_dev->rca_binaryname); + ret = -EINVAL; + goto out; + } + + mutex_lock(&tas_dev->pde_lock); + img_sz = fmw->size; + buf = fmw->data; + offset += FW_DL_OFFSET; + while (offset < (img_sz - FW_FL_HDR)) { + memset(&hdr, 0, sizeof(hdr)); + offset += read_header(&buf[offset], &hdr); + dev_dbg(tas_dev->dev, + "vndr=%d, file=%d, version=%d, len=%d, off=%d\n", + hdr.vendor_id, hdr.file_id, hdr.version, + hdr.length, offset); + /* size also includes the header */ + file_blk_size = hdr.length - FW_FL_HDR; + + switch (hdr.file_id) { + case 0: + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + PRAM_ADDR_START, file_blk_size, + &buf[offset]); + if (ret < 0) + dev_err(tas_dev->dev, + "PRAM update failed: %d", ret); + break; + + case 1: + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + YRAM_ADDR_START, file_blk_size, + &buf[offset]); + if (ret < 0) + dev_err(tas_dev->dev, + "YRAM update failed: %d", ret); + + break; + + default: + ret = -EINVAL; + dev_err(tas_dev->dev, "Unsupported file"); + break; + } + + if (ret == 0) + offset += file_blk_size; + else + break; + }; + mutex_unlock(&tas_dev->pde_lock); + tas2783_update_calibdata(tas_dev); + +out: + if (!ret) + tas_dev->fw_dl_success = true; + tas_dev->fw_dl_task_done = true; + wake_up(&tas_dev->fw_wait); + if (fmw) + release_firmware(fmw); +} + +static inline s32 tas_clear_latch(struct tas2783_prv *priv) +{ + return regmap_update_bits(priv->regmap, + TASDEV_REG_SDW(0, 0, 0x5c), + 0x04, 0x04); +} + +static s32 tas_fu21_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, s32 event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct tas2783_prv *tas_dev = snd_soc_component_get_drvdata(component); + s32 mute; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + mute = 1; + break; + } + + return sdw_write_no_pm(tas_dev->sdw_peripheral, + SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 1), mute); +} + +static s32 tas_fu23_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, s32 event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct tas2783_prv *tas_dev = snd_soc_component_get_drvdata(component); + s32 mute; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + mute = 1; + break; + } + + return sdw_write_no_pm(tas_dev->sdw_peripheral, + SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 1), mute); +} + +static const struct snd_soc_dapm_widget tas_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_DAC_E("FU21", NULL, SND_SOC_NOPM, 0, 0, tas_fu21_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("FU23", NULL, SND_SOC_NOPM, 0, 0, tas_fu23_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tas_audio_map[] = { + {"FU21", NULL, "ASI"}, + {"SPK", NULL, "FU21"}, + {"FU23", NULL, "ASI"}, + {"SPK", NULL, "FU23"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static s32 tas_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, s32 direction) +{ + if (!sdw_stream) + return 0; + + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tas_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static s32 tas_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + struct sdw_slave *sdw_peripheral = tas_dev->sdw_peripheral; + s32 ret, retry = 3; + + if (!tas_dev->fw_dl_success) { + dev_err(tas_dev->dev, "error playback without fw download"); + return -EINVAL; + } + + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + if (!sdw_stream) + return -EINVAL; + + ret = tas_clear_latch(tas_dev); + if (ret) + dev_err(tas_dev->dev, + "clear latch failed, err=%d", ret); + + mutex_lock(&tas_dev->pde_lock); + /* + * Sometimes, there is error returned during power on. + * So added retry logic to ensure power on so that + * port prepare succeeds + */ + do { + ret = regmap_write(tas_dev->regmap, + SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POW_STATE, 0), + TAS2783_SDCA_POW_STATE_ON); + if (!ret) + break; + usleep_range(2000, 2200); + } while (retry--); + mutex_unlock(&tas_dev->pde_lock); + if (ret) + return ret; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) + dev_err(dai->dev, "Unable to configure port\n"); + + return ret; +} + +static s32 tas_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + s32 ret; + struct snd_soc_component *component = dai->component; + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + sdw_stream_remove_slave(tas_dev->sdw_peripheral, sdw_stream); + + mutex_lock(&tas_dev->pde_lock); + ret = regmap_write(tas_dev->regmap, + SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POW_STATE, 0), + TAS2783_SDCA_POW_STATE_OFF); + mutex_unlock(&tas_dev->pde_lock); + + return ret; +} + +static const struct snd_soc_dai_ops tas_dai_ops = { + .hw_params = tas_sdw_hw_params, + .hw_free = tas_sdw_pcm_hw_free, + .set_stream = tas_set_sdw_stream, + .shutdown = tas_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tas_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tas_dai_ops, + .symmetric_rate = 1, + }, +}; + +static s32 tas_component_probe(struct snd_soc_component *component) +{ + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(component); + + tas_dev->component = component; + tas25xx_register_misc(tas_dev->sdw_peripheral); + + return 0; +} + +static void tas_component_remove(struct snd_soc_component *codec) +{ + struct tas2783_prv *tas_dev = + snd_soc_component_get_drvdata(codec); + tas25xx_deregister_misc(); + tas_dev->component = NULL; +} + +static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { + .probe = tas_component_probe, + .remove = tas_component_remove, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tas_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas_dapm_widgets), + .dapm_routes = tas_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static s32 tas_init(struct tas2783_prv *tas_dev) +{ + s32 ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tas_dai_driver, + ARRAY_SIZE(tas_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return ret; +} + +static s32 tas_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + s32 nval; + s32 i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = false; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = false; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 200; + + return 0; +} + +static s32 tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tas2783_prv *tas_dev = dev_get_drvdata(dev); + + if (!tas_dev->hw_init) + return 0; + + regcache_cache_only(tas_dev->regmap, true); + return 0; +} + +static s32 tas2783_sdca_dev_system_suspend(struct device *dev) +{ + return tas2783_sdca_dev_suspend(dev); +} + +static s32 tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tas2783_prv *tas_dev = dev_get_drvdata(dev); + unsigned long t; + + if (!slave->unattach_request) + goto regmap_sync; + + t = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT)); + if (!t) { + dev_err(&slave->dev, "resume: initialization timed out\n"); + sdw_show_ping_status(slave->bus, true); + return -ETIMEDOUT; + } + + slave->unattach_request = 0; + +regmap_sync: + regcache_cache_only(tas_dev->regmap, false); + regcache_sync(tas_dev->regmap); + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_system_suspend, tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, tas2783_sdca_dev_resume, NULL) +}; + +static s32 tas_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tas2783_prv *tas_dev = dev_get_drvdata(dev); + s32 ret; + u8 unique_id = tas_dev->sdw_peripheral->id.unique_id; + + if (tas_dev->hw_init) + return 0; + + tas_dev->fw_dl_task_done = false; + tas_dev->fw_dl_success = false; + scnprintf(tas_dev->rca_binaryname, sizeof(tas_dev->rca_binaryname), + "tas2783-%01x.bin", unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, + GFP_KERNEL, tas_dev, tas2783_fw_ready); + if (ret) { + dev_err(tas_dev->dev, + "firmware request failed for uid=%d, ret=%d\n", + unique_id, ret); + return ret; + } + + ret = wait_event_timeout(tas_dev->fw_wait, tas_dev->fw_dl_task_done, + msecs_to_jiffies(TIMEOUT_FW_DL_MS)); + if (!ret) { + dev_err(tas_dev->dev, "fw request, wait_event timeout\n"); + ret = -EAGAIN; + } else { + ret = regmap_multi_reg_write(tas_dev->regmap, tas2783_init_seq, + ARRAY_SIZE(tas2783_init_seq)); + tas_dev->hw_init = true; + } + + return ret; +} + +static s32 tas_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tas2783_prv *tas_dev = dev_get_drvdata(&slave->dev); + struct device *dev = &slave->dev; + + dev_dbg(dev, "Perifpheral status = %s", + status == SDW_SLAVE_UNATTACHED ? "unattached" : + status == SDW_SLAVE_ATTACHED ? "attached" : "alert"); + + tas_dev->status = status; + if (status == SDW_SLAVE_UNATTACHED) + tas_dev->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_dev->hw_init || tas_dev->status != SDW_SLAVE_ATTACHED) + return 0; + + /* updated the cache data to device */ + regcache_cache_only(tas_dev->regmap, false); + regcache_sync(tas_dev->regmap); + + /* perform I/O transfers required for Slave initialization */ + return tas_io_init(&slave->dev, slave); +} + +static const struct sdw_slave_ops tas_sdw_ops = { + .read_prop = tas_read_prop, + .update_status = tas_update_status, +}; + +static void tas_remove(struct tas2783_prv *tas_dev) +{ + snd_soc_unregister_component(tas_dev->dev); +} + +static s32 tas_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + struct device *dev = &peripheral->dev; + struct tas2783_prv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return dev_err_probe(dev, -ENOMEM, + "Failed devm_kzalloc"); + + tas_dev->dev = dev; + tas_dev->sdw_peripheral = peripheral; + tas_dev->hw_init = false; + mutex_init(&tas_dev->calib_lock); + mutex_init(&tas_dev->pde_lock); + + init_waitqueue_head(&tas_dev->fw_wait); + dev_set_drvdata(dev, tas_dev); + regmap = devm_regmap_init_sdw_mbq_cfg(peripheral, + &tas_regmap, + &tas2783_mbq_cfg); + if (IS_ERR(regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + /* keep in cache until the device is fully initialized */ + regcache_cache_only(regmap, true); + tas_dev->regmap = regmap; + return tas_init(tas_dev); +} + +static s32 tas_sdw_remove(struct sdw_slave *peripheral) +{ + struct tas2783_prv *tas_dev = dev_get_drvdata(&peripheral->dev); + + pm_runtime_disable(tas_dev->dev); + tas_remove(tas_dev); + mutex_destroy(&tas_dev->calib_lock); + mutex_destroy(&tas_dev->pde_lock); + dev_set_drvdata(&peripheral->dev, NULL); + + return 0; +} + +static const struct sdw_device_id tas_sdw_id[] = { + /* chipid for the TAS2783 is 0x0000 */ + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, tas_sdw_id); + +static struct sdw_driver tas_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = pm_ptr(&tas2783_sdca_pm), + }, + .probe = tas_sdw_probe, + .remove = tas_sdw_remove, + .ops = &tas_sdw_ops, + .id_table = tas_sdw_id, +}; +module_sdw_driver(tas_sdw_driver); + +MODULE_AUTHOR("Texas Instruments Inc."); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..a77e332dee3a --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,107 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2025 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2783 driver implements a flexible and configurable + * algo coefficient setting for single TAS2783 chips. + * + * Author: Niranjan H Y <niranjanhy(a)ti.com> + * Author: Baojun Xu <baojun.xu(a)ti.com> + */ +#include <linux/workqueue.h> + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* book, page, register */ +#define TASDEV_REG_SDW(book, page, reg) (((book) * 256 * 128) + \ + 0x800000 + ((page) * 128) + (reg)) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEV_REG_SDW(0x0, 0x00, 0x1A) +#define TAS2783_AMP_LEVEL TASDEV_REG_SDW(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +#define PRAM_ADDR_START TASDEV_REG_SDW(0x8c, 0x01, 0x8) +#define PRAM_ADDR_END TASDEV_REG_SDW(0x8c, 0xff, 0x7f) +#define YRAM_ADDR_START TASDEV_REG_SDW(0x00, 0x02, 0x8) +#define YRAM_ADDR_END TASDEV_REG_SDW(0x00, 0x37, 0x7f) + +/* Calibration data */ +#define TAS2783_CAL_R0 TASDEV_REG_SDW(0, 0x16, 0x4C) +#define TAS2783_CAL_INVR0 TASDEV_REG_SDW(0, 0x16, 0x5C) +#define TAS2783_CAL_R0LOW TASDEV_REG_SDW(0, 0x16, 0x64) +#define TAS2783_CAL_POWER TASDEV_REG_SDW(0, 0x15, 0x44) +#define TAS2783_CAL_TLIM TASDEV_REG_SDW(0, 0x17, 0x58) + +/* TAS2783 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ + +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 +#define TAS2783_SDCA_ENT_FU26 0x03 +#define TAS2783_SDCA_ENT_XU22 0x04 +#define TAS2783_SDCA_ENT_CS24 0x05 +#define TAS2783_SDCA_ENT_CS21 0x06 +#define TAS2783_SDCA_ENT_CS25 0x07 +#define TAS2783_SDCA_ENT_CS26 0x08 +#define TAS2783_SDCA_ENT_CS28 0x09 +#define TAS2783_SDCA_ENT_PDE23 0x0C +#define TAS2783_SDCA_ENT_UDMPU23 0x0E +#define TAS2783_SDCA_ENT_SAPU29 0x0F +#define TAS2783_SDCA_ENT_PPU21 0x10 +#define TAS2783_SDCA_ENT_PPU26 0x11 +#define TAS2783_SDCA_ENT_TG23 0x12 +#define TAS2783_SDCA_ENT_IT21 0x13 +#define TAS2783_SDCA_ENT_IT29 0x14 +#define TAS2783_SDCA_ENT_IT26 0x15 +#define TAS2783_SDCA_ENT_IT28 0x16 +#define TAS2783_SDCA_ENT_OT24 0x17 +#define TAS2783_SDCA_ENT_OT23 0x18 +#define TAS2783_SDCA_ENT_OT25 0x19 +#define TAS2783_SDCA_ENT_OT28 0x1A +#define TAS2783_SDCA_ENT_MU26 0x1b +#define TAS2783_SDCA_ENT_OT127 0x1E +#define TAS2783_SDCA_ENT_FU127 0x1F +#define TAS2783_SDCA_ENT_CS127 0x20 +#define TAS2783_SDCA_ENT_MFPU21 0x22 +#define TAS2783_SDCA_ENT_MFPU26 0x23 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POW_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 + +#define TAS2783_SDCA_POW_STATE_ON 0 +#define TAS2783_SDCA_POW_STATE_OFF 3 + +/* calibration data */ +#define TAS2783_CALIB_PARAMS 6 /* 5 + 1 unique id */ +#define TAS2783_CALIB_MAX_SPK_COUNT 8 +#define TAS2783_CALIB_HDR_SZ 12 +#define TAS2783_CALIB_CRC_SZ 4 +#define TAS2783_CALIB_DATA_SZ ((TAS2783_CALIB_HDR_SZ) + TAS2783_CALIB_CRC_SZ + \ + ((TAS2783_CALIB_PARAMS) * 4 * (TAS2783_CALIB_MAX_SPK_COUNT))) + +#if IS_ENABLED(CONFIG_SND_SOC_TAS2783_UTIL) +int32_t tas25xx_register_misc(struct sdw_slave *peripheral); +int32_t tas25xx_deregister_misc(void); +#endif + +#endif /*__TAS2783_H__ */ -- 2.25.1

2 4

Segfault in snd_config_delete due to incorrect node deletion in parse_array_def / parse_def
by GitHub issues - opened 09 Sep '25

09 Sep '25

alsa-project/alsa-lib issue #477 was opened from midzishi: While fuzzing snd_config_load() a reproducible segmentation fault inside snd_config_delete() was discovered. The root cause is incorrect cleanup logic in the configuration parser functions (parse_array_def() and parse_def()). In multiple branches, existing configuration nodes are replaced incorrectly: - The parser sometimes calls snd_config_delete(n) where n is NULL or a newly allocated node, instead of deleting the existing node returned by _snd_config_search() (variable g). - In other cases, a newly created node is added to the parent before its contents are successfully parsed. If parsing fails, the rollback path deletes this node explicitly, and later the parent deletes it again during recursive cleanup, so it causes double-free or corrupted linked list. This results in corrupted parent–child relationships and eventually SEGV when snd_config_delete() tries to read config->refcount of an already freed node. In both parse_array_def() and parse_def(), _snd_config_make_add() is called too early (before parsing child contents). On error, the parser deletes the child manually, but the parent will later delete it again during recursive cleanup. Proposed fix: ``` diff --git a/src/conf.c b/src/conf.c index 905c8f4d..7e403192 100644 --- a/src/conf.c +++ b/src/conf.c @@ -1274,7 +1274,8 @@ static int parse_array_def(snd_config_t *parent, input_t *input, int *idx, int s snprintf(static_id, sizeof(static_id), "%i", *idx); if (_snd_config_search(parent, static_id, -1, &g) == 0) { if (override) { - snd_config_delete(n); + snd_config_delete(g); + g = NULL; } else { /* merge */ (*idx)++; @@ -1305,7 +1306,7 @@ static int parse_array_def(snd_config_t *parent, input_t *input, int *idx, int s goto __end; } } else { - err = _snd_config_make_add(&n, &id, SND_CONFIG_TYPE_COMPOUND, parent); + err = _snd_config_make(&n, &id, SND_CONFIG_TYPE_COMPOUND); if (err < 0) goto __end; } @@ -1328,6 +1329,10 @@ static int parse_array_def(snd_config_t *parent, input_t *input, int *idx, int s err = LOCAL_UNEXPECTED_CHAR; goto __end; } + if (!skip && n && !n->parent) { + err = snd_config_add(parent, n); + if (err < 0) { snd_config_delete(n); goto __end; } + } break; } default: @@ -1422,7 +1427,7 @@ static int parse_def(snd_config_t *parent, input_t *input, int skip, int overrid err = -ENOENT; goto __end; } - err = _snd_config_make_add(&n, &id, SND_CONFIG_TYPE_COMPOUND, parent); + err = _snd_config_make(&n, &id, SND_CONFIG_TYPE_COMPOUND); if (err < 0) goto __end; n->u.compound.join = true; @@ -1464,7 +1469,7 @@ static int parse_def(snd_config_t *parent, input_t *input, int skip, int overrid goto __end; } } else { - err = _snd_config_make_add(&n, &id, SND_CONFIG_TYPE_COMPOUND, parent); + err = _snd_config_make(&n, &id, SND_CONFIG_TYPE_COMPOUND); if (err < 0) goto __end; } @@ -1483,6 +1488,10 @@ static int parse_def(snd_config_t *parent, input_t *input, int skip, int overrid err = LOCAL_UNEXPECTED_CHAR; goto __end; } + if (!skip && n && !n->parent) { + err = snd_config_add(parent, n); + if (err < 0) { snd_config_delete(n); goto __end; } + } break; } default: ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/477 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v4 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 09 Sep '25

09 Sep '25

Add tas2020, tas2118, tas2120, tas2320, tas2570, tas2572, tas5825 tas5827 support in tas2781 driver. Tas2118, tas2x20, tas257x have no on-chip DSP, tas582x have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v4: - Change description for adding tas2570, tas2572 and tas5827 - Add tas2570, tas2572, tas5827 in device enumerate - Change the check for reset value, from "== TAS5825" to ">= TAS5825" as TAS5827 was added. - Add tas2570, tas2572, tas5827 in i2c_device_id list - Add tas2570, tas2572, tas5827 in of_device_id list - Change max of device ID from TAS2781 to TAS_OTHERS - Change the check for non-DSP chip, from "< TAS2563" to switch case, as TAS2570 and TAS2572 also have no on-chip DSP - Add TAS5827 in comments for calibration check - Change the check for on-chip DSP, from ">= TAS2563" to switch case, as TAS2570 and TAS2572 also have no on-chip DSP - Add tas2570, tas2572, tas5827 initialyze in codec probe - Add tas2570, tas2572, tas5827 in ACPI device id list v3: - No update. v2: - Update the mail list for maintainers of yaml file --- include/sound/tas2781.h | 14 +- include/sound/tas2x20-tlv.h | 259 ++++++++++++++++++++++++++ sound/soc/codecs/tas2781-comlib-i2c.c | 2 + sound/soc/codecs/tas2781-i2c.c | 184 +++++++++++++----- 4 files changed, 410 insertions(+), 49 deletions(-) create mode 100644 include/sound/tas2x20-tlv.h diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index f0aefc04a957..ddd997ac3216 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -51,7 +51,9 @@ /* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) + +#define TAS5825_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) @@ -110,8 +112,17 @@ #define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { + TAS2020, + TAS2118, + TAS2120, + TAS2320, TAS2563, + TAS2570, + TAS2572, TAS2781, + TAS5825, + TAS5827, + TAS_OTHERS, }; enum dspbin_type { @@ -194,6 +205,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const unsigned char (*dvc_tlv_table)[4]; const char *name_prefix; unsigned char ndev; unsigned int dspbin_typ; diff --git a/include/sound/tas2x20-tlv.h b/include/sound/tas2x20-tlv.h new file mode 100644 index 000000000000..6e6bcec4a0a1 --- /dev/null +++ b/include/sound/tas2x20-tlv.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2x20/TAS2118 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2x20/TAS2118 hda driver implements for one, two, or even multiple +// TAS2x20/TAS2118 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2X20_TLV_H__ +#define __TAS2X20_TLV_H__ + +#define TAS2X20_DVC_LEVEL TASDEVICE_REG(0x0, 0x2, 0x0c) +#define TAS2X20_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x07) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_amp_tlv, 2100, 50, 0); + +/* pow(10, db/20) * pow(2,22) */ +static const __maybe_unused unsigned char tas2x20_dvc_table[][4] = { + { 0X00, 0X00, 0X0D, 0X00 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0X00 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0X00 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0X00 }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0X00 }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0X00 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X00 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X00 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X00 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0X00 }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X00 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X00 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0X00 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X00 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X00 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X00 }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X00 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X00 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0X00 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X00 }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X00 }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0X00 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0X00 }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0X00 }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X00 }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0X00 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X00 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X00 }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X00 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X00 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0X00 }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0X00 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0X00 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X00 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X00 }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X00 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X00 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X00 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0X00 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X00 }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0X00 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0X00 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0X00 }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0X00 }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X00 }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X00 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X00 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0X00 }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0X00 }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0X00 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0X00 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X00 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0X00 }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X00 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X00 }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0X00 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0X00 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0X00 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X00 }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X00 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X00 }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X00 }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X00 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X00 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X00 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X00 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0X00 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X00 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0X00 }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X00 }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0X00 }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X00 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0X00 }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0X00 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X00 }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0X00 }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X00 }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X00 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X00 }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0X00 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X00 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X00 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0X00 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0X00 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X00 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X00 }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X00 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0X00 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0X00 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X00 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X00 }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X00 }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X00 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X00 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0X00 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X00 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X00 }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0X00 }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X00 }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0X00 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X00 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0X00 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X00 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X00 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X00 }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0X00 }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X00 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X00 }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0X00 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X00 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0X00 }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X00 }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0X00 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X00 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X00 }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0X00 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X00 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X00 }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X00 }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X00 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0X00 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0X00 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0X00 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X00 }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X00 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X00 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X00 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0X00 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X00 }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0X00 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X00 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X00 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X00 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0X00 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X00 }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0X00 }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0X00 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0X00 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X00 }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X00 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0X00 }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X00 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X00 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0X00 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X00 }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0X00 }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X00 }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X00 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X00 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0X00 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X00 }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X00 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0X00 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X00 }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X00 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0X00 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X00 }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X00 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X00 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0X00 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X00 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0X00 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X00 }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X00 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0X00 }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X00 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X00 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X00 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X00 }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0X00 }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0X00 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0X00 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0X00 }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X00 }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X00 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0X00 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0X00 }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0X00 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X00 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X00 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0X00 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0X00 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X00 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X00 }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X00 }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0X00 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X00 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X00 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X00 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0X00 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0X00 }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X00 }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0X00 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X00 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X00 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0X00 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X00 }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X00 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X00 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0X00 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X00 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X00 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X00 }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X00 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0X00 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X00 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X00 }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0X00 }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X00 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X00 }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0X00 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X00 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0X00 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X00 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X00 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X00 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X00 }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X00 }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X00 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X00 }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0X00 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0X00 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X00 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0X00 }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X00 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0X00 }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X00 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X00 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0X00 }, /* 5.5db */ + { 0X7F, 0XB2, 0X61, 0X00 }, /* 6.0db */ +}; +#endif diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c index c078bb0a8437..b3fd7350143b 100644 --- a/sound/soc/codecs/tas2781-comlib-i2c.c +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -320,6 +320,8 @@ void tasdevice_reset(struct tasdevice_priv *tas_dev) for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, TASDEVICE_REG_SWRESET, + tas_dev->chip_id >= TAS5825 ? + TAS5825_REG_SWRESET_RESET : TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index ea3cdb8553de..1539b70881d1 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,8 +30,10 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2x20-tlv.h> #include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include <linux/unaligned.h> #define X2563_CL_STT_VAL(xreg, xval) \ @@ -98,16 +100,32 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { }; static const struct i2c_device_id tasdevice_id[] = { + { "tas2020", TAS2020 }, + { "tas2118", TAS2118 }, + { "tas2120", TAS2120 }, + { "tas2320", TAS2320 }, { "tas2563", TAS2563 }, + { "tas2570", TAS2570 }, + { "tas2572", TAS2572 }, { "tas2781", TAS2781 }, + { "tas5825", TAS5825 }, + { "tas5827", TAS5827 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); #ifdef CONFIG_OF static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2020" }, + { .compatible = "ti,tas2118" }, + { .compatible = "ti,tas2120" }, + { .compatible = "ti,tas2320" }, { .compatible = "ti,tas2563" }, + { .compatible = "ti,tas2570" }, + { .compatible = "ti,tas2572" }, { .compatible = "ti,tas2781" }, + { .compatible = "ti,tas5825" }, + { .compatible = "ti,tas5827" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -797,7 +815,7 @@ static int tasdev_nop_get( return 0; } -static int tas2563_digital_gain_get( +static int tasdevice_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -823,15 +841,15 @@ static int tas2563_digital_gain_get( while (r > 1 + l) { mid = (l + r) / 2; - ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + ar_mid = get_unaligned_be32(tas_dev->dvc_tlv_table[mid]); if (target < ar_mid) r = mid; else l = mid; } - ar_l = get_unaligned_be32(tas2563_dvc_table[l]); - ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + ar_l = get_unaligned_be32(tas_dev->dvc_tlv_table[l]); + ar_r = get_unaligned_be32(tas_dev->dvc_tlv_table[r]); /* find out the member same as or closer to the current volume */ ucontrol->value.integer.value[0] = @@ -841,7 +859,7 @@ static int tas2563_digital_gain_get( return 0; } -static int tas2563_digital_gain_put( +static int tasdevice_digital_gain_put( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -867,7 +885,7 @@ static int tas2563_digital_gain_put( } volrd = get_unaligned_be32(&data[0]); - volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + volwr = get_unaligned_be32(tas_dev->dvc_tlv_table[vol]); if (volrd == volwr) { rc = 0; @@ -876,7 +894,7 @@ static int tas2563_digital_gain_put( for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_bulk_write(tas_dev, i, reg, - (unsigned char *)tas2563_dvc_table[vol], 4); + (unsigned char *)tas_dev->dvc_tlv_table[vol], 4); if (ret) { dev_err(tas_dev->dev, "%s, set digital vol error in dev %d\n", @@ -892,11 +910,6 @@ static int tas2563_digital_gain_put( return rc; } -static const struct snd_kcontrol_new tasdevice_snd_controls[] = { - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tasdev_force_fwload_get, tasdev_force_fwload_put), -}; - static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, tasdev_nop_get, tasdev_calib_stop_put), @@ -907,6 +920,16 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), }; +static const struct snd_kcontrol_new tas2x20_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2X20_AMP_LEVEL, + 0, 0, 42, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas2x20_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2X20_DVC_LEVEL, + 0, 0, ARRAY_SIZE(tas2x20_dvc_table) - 1, 0, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, + tas2x20_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -916,6 +939,15 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, tas2781_dvc_tlv), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_cali_controls[] = { SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), }; @@ -923,7 +955,7 @@ static const struct snd_kcontrol_new tas2781_cali_controls[] = { static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, - tas2563_digital_gain_get, tas2563_digital_gain_put, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, tas2563_dvc_tlv), }; @@ -968,8 +1000,8 @@ static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; - uinfo->value.integer.min = TAS2563; - uinfo->value.integer.max = TAS2781; + uinfo->value.integer.min = TAS2020; + uinfo->value.integer.max = TAS_OTHERS; return 0; } @@ -1168,9 +1200,9 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *active_dev_num, *chip_id; + char *active_dev_num, *chip_id, *fw_load; char *conf_name, *prog_name; - int nr_controls = 4; + int nr_controls = 5; int mix_index = 0; /* Alloc kcontrol via devm_kzalloc, which don't manually @@ -1228,6 +1260,19 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; + fw_load = devm_kstrdup(tas_priv->dev, "Speaker Force Firmware Load", + GFP_KERNEL); + if (!fw_load) + return -ENOMEM; + + dsp_ctrls[mix_index].name = fw_load; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_info_bool_ext; + dsp_ctrls[mix_index].put = tasdev_force_fwload_put; + dsp_ctrls[mix_index].get = tasdev_force_fwload_get; + dsp_ctrls[mix_index].private_value = 0UL; + mix_index++; + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); } @@ -1587,6 +1632,16 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + /* There is no DSP firmware required for TAS2118/2X20/257X. */ + switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + case TAS2570: + case TAS2572: + goto out; + } if (tas_priv->name_prefix) scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); @@ -1608,34 +1663,37 @@ static void tasdevice_fw_ready(const struct firmware *fmw, dev_err(tas_priv->dev, "dsp controls error\n"); goto out; } - - ret = tasdevice_create_cali_ctrls(tas_priv); - if (ret) { - dev_err(tas_priv->dev, "cali controls error\n"); - goto out; - } - tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - /* If calibrated data occurs error, dsp will still works with default - * calibrated data inside algo. - */ - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->name_prefix) - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, - tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - else - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s_cal_0x%02x.bin", tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - ret = tas2781_load_calibration(tas_priv, - tas_priv->cal_binaryname[i], i); - if (ret != 0) - dev_err(tas_priv->dev, - "%s: load %s error, default will effect\n", - __func__, tas_priv->cal_binaryname[i]); + /* There is no calibration required for TAS5825/TAS5827. */ + if (tas_priv->chip_id < TAS5825) { + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + /* If calibrated data occurs error, dsp will still works + * with default calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", + tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, keep default.\n", + __func__, tas_priv->cal_binaryname[i]); + } } tasdevice_prmg_load(tas_priv, 0); @@ -1659,8 +1717,14 @@ static void tasdevice_fw_ready(const struct firmware *fmw, #endif out: if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { - /* If DSP FW fail, DSP kcontrol won't be created. */ - tasdevice_dsp_remove(tas_priv); + switch (tas_priv->chip_id) { + case TAS2563: + case TAS2781: + case TAS5825: + case TAS5827: + /* If DSP FW fail, DSP kcontrol won't be created. */ + tasdevice_dsp_remove(tas_priv); + } } mutex_unlock(&tas_priv->codec_lock); release_firmware(fmw); @@ -1804,13 +1868,30 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) int rc; switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + case TAS2570: + case TAS2572: + p = (struct snd_kcontrol_new *)tas2x20_snd_controls; + size = ARRAY_SIZE(tas2x20_snd_controls); + tas_priv->dvc_tlv_table = tas2x20_dvc_table; + break; case TAS2781: p = (struct snd_kcontrol_new *)tas2781_snd_controls; size = ARRAY_SIZE(tas2781_snd_controls); break; + case TAS5825: + case TAS5827: + p = (struct snd_kcontrol_new *)tas5825_snd_controls; + size = ARRAY_SIZE(tas5825_snd_controls); + break; default: p = (struct snd_kcontrol_new *)tas2563_snd_controls; size = ARRAY_SIZE(tas2563_snd_controls); + tas_priv->dvc_tlv_table = tas2563_dvc_table; + break; } rc = snd_soc_add_component_controls(codec, p, size); @@ -1850,8 +1931,6 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tasdevice_snd_controls, - .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, @@ -1967,7 +2046,16 @@ static void tasdevice_i2c_remove(struct i2c_client *client) #ifdef CONFIG_ACPI static const struct acpi_device_id tasdevice_acpi_match[] = { - { "TAS2781", TAS2781 }, + { "TXNW2020", TAS2020 }, + { "TXNW2118", TAS2118 }, + { "TXNW2120", TAS2120 }, + { "TXNW2320", TAS2320 }, + { "TXNW2563", TAS2563 }, + { "TXNW2570", TAS2570 }, + { "TXNW2572", TAS2572 }, + { "TXNW2781", TAS2781 }, + { "TXNW5825", TAS5825 }, + { "TXNW5827", TAS5827 }, {}, }; -- 2.43.0

4 6

ASRock X870 Riptide: extremely low mic capture volume
by GitHub issues - opened 09 Sep '25

09 Sep '25

alsa-project/alsa-ucm-conf issue #611 was opened from jktomer: I'm running with an ASRock X870 Riptide Wifi, which seems to use the same ALC4082 as other ASRock socket AM5 boards: ``` :) 18:00 viola:jkt[3828]alsa% lsusb | grep audio Bus 001 Device 008: ID 26ce:0a08 Generic USB Audio ``` Playback on this device seems to work and the various ports are recognized apparently correctly, but capture from both the front microphone and rear line-input ports is excessively low volume even with the mixer turned up as high as it goes. (The card does still show in `/proc/asound/cards` as "Generic USB Audio", but editing the relevant section in `ALC4080-HiFi.conf` seems to influence the output of `alsaucm -c hw:1 dump text`, so I'm assuming the recognition part works correctly.) Output of `alsa-info.sh` and `alsaucm -c hw:1 dump text` attached. [alsa-info.txt](https://github.com/user-attachments/files/22223994/alsa-info… [alsaucm.txt](https://github.com/user-attachments/files/22223995/alsaucm.txt) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/611 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM2: Intel: sof-hda-dsp: HiFi: IPC3 mono DMIC is exposed as stereo PCM
by GitHub pull_request - opened 08 Sep '25

08 Sep '25

alsa-project/alsa-ucm-conf pull request #610 was opened from ujfalusi: With IPC3 the DMIC PCM is stereo even in case of a single (mono) DMIC connection. Make the `CaptureChannels 1` conditional to IPC4 only. Closes: https://bugzilla.redhat.com/show_bug.cgi?id=2393552 Closes: https://bugzilla.redhat.com/show_bug.cgi?id=2384324 Closes: https://github.com/thesofproject/linux/issues/5528 Fixes: 56cbdfd04339 ("UCM2: Intel: sof-hda-dsp: HiFi: Fix handling of mono DMICs") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/610 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/610.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4] ALSA: hda/tas2781: Fix the order of TAS2781 calibrated-data
by Shenghao Ding 08 Sep '25

08 Sep '25

A bug reported by one of my customers that the order of TAS2781 calibrated-data is incorrect, the correct way is to move R0_Low and insert it between R0 and InvR0. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - Add missing base into cali_cnv(). v3: - Take Tiwai's advice on cali_cnv() to make it more simpler. v2: - Submit to the sound branch maintained by Tiwai instead of linux-next branch - Drop other fix --- sound/hda/codecs/side-codecs/tas2781_hda.c | 25 +++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index f46d2e06c64f..536940c78f00 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -33,6 +33,23 @@ const efi_guid_t tasdev_fct_efi_guid[] = { }; EXPORT_SYMBOL_NS_GPL(tasdev_fct_efi_guid, "SND_HDA_SCODEC_TAS2781"); +/* + * The order of calibrated-data writing function is a bit different from the + * order in UEFI. Here is the conversion to match the order of calibrated-data + * writing function. + */ +static void cali_cnv(unsigned char *data, unsigned int base, int offset) +{ + struct cali_reg reg_data; + + memcpy(&reg_data, &data[base], sizeof(reg_data)); + /* the data order has to be swapped between r0_low_reg and inv0_reg */ + swap(reg_data.r0_low_reg, reg_data.invr0_reg); + + cpu_to_be32_array((__force __be32 *)(data + offset + 1), + (u32 *)&reg_data, TASDEV_CALIB_N); +} + static void tas2781_apply_calib(struct tasdevice_priv *p) { struct calidata *cali_data = &p->cali_data; @@ -103,8 +120,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) data[l] = k; oft++; - for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i + 1] = data[4 * oft + i]; + cali_cnv(data, 4 * oft, l); k++; } } @@ -130,9 +146,8 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) for (j = p->ndev - 1; j >= 0; j--) { l = j * (cali_data->cali_dat_sz_per_dev + 1); - for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) - data[l + i] = data[p->index * 5 + i]; - data[l+i] = j; + cali_cnv(data, cali_data->cali_dat_sz_per_dev * j, l); + data[l] = j; } } -- 2.43.0

2 1

Feature Request: conf.d for USB-Audio
by GitHub issues - edited 08 Sep '25

08 Sep '25

alsa-project/alsa-ucm-conf issue #609 was edited from FrostyCoolSlug: The release cadence of `alsa-ucm-conf` tends to be a little on the slow side (generally around 6 months between releases), but the problem I'm hitting currently is the requirement for users to manually edit `USB-Audio.conf` in order to get earlier support for their devices (and depending on distribution, support could be delayed upwards of two years). I think a method where a package could be created to drop files into a conf.d style directory would allow for a users device to be configured, without the need for manually editing `USB-Audio.conf` to add support. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/609 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Feature Request: conf.d for USB-Audio
by GitHub issues - opened 08 Sep '25

08 Sep '25

alsa-project/alsa-ucm-conf issue #609 was opened from FrostyCoolSlug: The release cadence of `alsa-ucm-conf` tends to be a little on the slow side (generally around 6 months between releases), but the problem I'm hitting currently is the requirement for users to manually edit `USB-Audio.conf` in order to get support for their devices (and depending on distribution, support could be delayed upwards of two years). I think a method where a package could be created to drop files into a conf.d style directory would allow for a users device to be configured, without the need for manually editing `USB-Audio.conf` to add support. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/609 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Mic from audio jack not work on Dell Precision M4800. Please add mic drivers
by GitHub issues - edited 07 Sep '25

07 Sep '25

alsa-project/alsa-lib issue #476 was edited from Slavomir9: Hello, on my Dell Precision M4800 not work mic on headset Koss CS100 audio jack (non USB) version. Please add mic drivers. [Dell Precision M4800 ](https://www.google.com/search?q=%0D%0ADell+Precision+M4800%0D%0A&sca_esv=e618e5e4341fb9f4&) [Headset Koss CS100](https://www.kaufland.sk/product/5373091/) I use dual boot. On Windows 11 work mic only after install drivers from Dell website. On Dell website are drivers for Linux for extremely old version of Ubuntu: Ubuntu® 12.04 More information about bug here https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/2122269 Please add mic drivers. Thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/476 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Mic from audio jack not work on Dell Precision M4800. Please add mic drivers
by GitHub issues - edited 07 Sep '25

07 Sep '25

alsa-project/alsa-lib issue #476 was edited from Slavomir9: Hello, on my Dell Precision M4800 not work mic on headset Koss CS100 audio jack (non USB) version. Please add mic drivers. [Dell Precision M4800 ](https://www.google.com/search?q=%0D%0ADell+Precision+M4800%0D%0A&sca_esv=e618e5e4341fb9f4&) [Headset Koss CS100](https://www.kaufland.sk/product/5373091/) I use dual boot. On Windows 11 work mic only after install drivers from Dell website. More information about bug here https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/2122269 Please add mic drivers. Thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/476 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Mic from audio jack not work on Dell Precision M4800. Please add mic drivers
by GitHub issues - edited 07 Sep '25

07 Sep '25

alsa-project/alsa-lib issue #476 was edited from Slavomir9: Hello, on my Dell Precision M4800 not work mic on headset Koss CS100 audio jack (non USB) version. Please add mic drivers. [Dell Precision M4800 ](https://www.google.com/search?q=%0D%0ADell+Precision+M4800%0D%0A&sca_esv=e618e5e4341fb9f4&) [Headset Koss CS100](https://www.kaufland.sk/product/5373091/) I use dual boot. On Windows 11 work mic only after install drivers from Dell website. Please add mic drivers. Thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/476 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Mic from audio jack not work on Dell Precision M4800
by GitHub issues - edited 07 Sep '25

07 Sep '25

alsa-project/alsa-lib issue #476 was edited from Slavomir9: Hello, on my Dell Precision M4800 not work mic on headset Koss CS100 audio jack (non USB) version. Please add mic drivers. [Dell Precision M4800 ](https://www.google.com/search?q=%0D%0ADell+Precision+M4800%0D%0A&sca_esv=e618e5e4341fb9f4&) [Headset Koss CS100](https://www.kaufland.sk/product/5373091/) I use dual boot. On Windows 11 work mic only after install drivers from Dell website. Please add mic drivers. Thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/476 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Mic from audio jack not work on Dell Precision M4800
by GitHub issues - opened 07 Sep '25

07 Sep '25

alsa-project/alsa-lib issue #476 was opened from Slavomir9: Hello, on my Dell Precision M4800 not work mic on headset Koss CS100 audio jack (non USB) version. [Dell Precision M4800 ](https://www.google.com/search?q=%0D%0ADell+Precision+M4800%0D%0A&sca_esv=e618e5e4341fb9f4&) [Headset Koss CS100](https://www.kaufland.sk/product/5373091/) I use dual boot. On Windows 11 work mic only after install drivers from Dell website Issue URL : https://github.com/alsa-project/alsa-lib/issues/476 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3] ALSA: hda/tas2781: Fix the order of TAS2781 calibrated-data
by Shenghao Ding 07 Sep '25

07 Sep '25

A bug reported by one of my customers that the order of TAS2781 calibrated-data is incorrect, the correct way is to move R0_Low and insert it between R0 and InvR0. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Take Tiwai's advice on cali_cnv() to make it more simpler. v2: - Submit to sound branch maintianed by Tiwai instead of linux-next branch - Drop other fix --- sound/hda/codecs/side-codecs/tas2781_hda.c | 25 +++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index f46d2e06c64f..f4a44c6b0234 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -33,6 +33,23 @@ const efi_guid_t tasdev_fct_efi_guid[] = { }; EXPORT_SYMBOL_NS_GPL(tasdev_fct_efi_guid, "SND_HDA_SCODEC_TAS2781"); +/* + * The order of calibrated-data writing function is a bit different from the + * order in UEFI. Here is the conversion to match the order of calibrated-data + * writing function. + */ +static void cali_cnv(unsigned char *data, unsigned int base, int offset) +{ + struct cali_reg reg_data; + + memcpy(&reg_data, data, sizeof(reg_data)); + /* the data order has to be swapped between r0_low_reg and inv0_reg */ + swap(reg_data.r0_low_reg, reg_data.invr0_reg); + + cpu_to_be32_array((__force __be32 *)(data + offset + 1), + (u32 *)&reg_data, TASDEV_CALIB_N); +} + static void tas2781_apply_calib(struct tasdevice_priv *p) { struct calidata *cali_data = &p->cali_data; @@ -103,8 +120,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) data[l] = k; oft++; - for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i + 1] = data[4 * oft + i]; + cali_cnv(data, 4 * oft, l); k++; } } @@ -130,9 +146,8 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) for (j = p->ndev - 1; j >= 0; j--) { l = j * (cali_data->cali_dat_sz_per_dev + 1); - for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) - data[l + i] = data[p->index * 5 + i]; - data[l+i] = j; + cali_cnv(data, cali_data->cali_dat_sz_per_dev * j, l); + data[l] = j; } } -- 2.43.0

3 2

No sound, only dummy output since updating alsa-lib to 1.2.14
by GitHub issues - opened 06 Sep '25

06 Sep '25

alsa-project/alsa-lib issue #475 was opened from aaaaadrien: Hello, Since an update on Fedora Linux 42, Audio card not detected, GNOME only shows "Dummy Output". No sound over speakers or HDMI. I downgraded the package alsa-lib (from alsa-lib-0:1.2.14-3.fc42.x86_64 to alsa-lib-0:1.2.13-4.fc42.x86_64) and now i have sound again. Tested with OpenSuse Tumbleweed Live ISO GNOME, with libasound2-1.2.14, i reproduce the issue. So i think it's not a distro issue but an ALSA Issue. That's why i open a bug here. Laptop model : Dell Latitude 3510 Audio Device : Intel Comet Lake PCH-LP cAVS driver: sof-audio-pci-intel-cnl API: ALSA version: k6.16.4-200.fc42.x86_64 status: kernel-api Reproducible: Always Steps to Reproduce: 1. Update alsa-lib to 1.2.14-3.fc42.x86_64 2. Reboot 3. See you only have dummy output Actual Results: No sound Expected Results: Sound always here ! Additional Information: Laptop model : Dell Latitude 3510 Audio Device : Intel Comet Lake PCH-LP cAVS driver: sof-audio-pci-intel-cnl API: ALSA version: k6.16.4-200.fc42.x86_64 status: kernel-api I opened a bug on Red Hat bugzilla yesterday but i enclose the link if you want to see the commands and checks i tested : https://bugzilla.redhat.com/show_bug.cgi?id=2393552 Issue URL : https://github.com/alsa-project/alsa-lib/issues/475 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSAfirewire broken
by M. Armsby 04 Sep '25

04 Sep '25

Hello everyone. 2 Firewire problems with ALSAfirewire. 1. Firewire Midi was working fine with kernel 5.10 with Reaper. Then obviously a lot changed with the Firewire Stack. The midi output from Reaper has not changed according to it's developer - Justin. The problem is direct ALSA-Midi. We tested the midi app from www.alsa-project.org/alsa-doc/alsa-lib/_2test_2rawmidi_8c-example.html#a0 yesterday and it freezes Debian and just breaks Arch with kernel 6.15 Thread at Reaper Bugs forum: https://forum.cockos.com/showthread.php?t=300278 Has low level ALSA-Midi API changed? 2. If using Pipewire-Jack with ALSAfirewire the Reaper midi output works fine but unfortunately Pipewire adds 90ms delay when playing back through any Firewire interface compared with an onboard soundcard. The pipewire dev Wim Tayman, says it's ALSAfirewire fault but I proved it is not - see thread: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues?show=eyJpaWQiOiI0… The ALSA recording and playback is confirmed by my tests to be working under 10ms RTT with Echo, Motu and Bebop interfaces. Is this problem related to first? Low level communication error? Cheers - and let me know if I can help in any way, although I'm not a programmer. P. S. I hope you can read my post as I loaded up K-9 eMail client so as to deliver text only. -- Martin Armsby

3 11

[PATCH v2] ALSA: hda/tas2781: Fix the order of TAS2781 calibrated-data
by Shenghao Ding 04 Sep '25

04 Sep '25

A bug reported by one of my customers that the order of TAS2781 calibrated-data is incorrect, the correct way is to move R0_Low and insert it between R0 and InvR0. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Submit to sound branch maintianed by Tiwai instead of linux-next branch - drop other fix --- sound/hda/codecs/side-codecs/tas2781_hda.c | 38 ++++++++++++++++++---- 1 file changed, 32 insertions(+), 6 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index f46d2e06c64f..cd9990869e18 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -33,6 +33,32 @@ const efi_guid_t tasdev_fct_efi_guid[] = { }; EXPORT_SYMBOL_NS_GPL(tasdev_fct_efi_guid, "SND_HDA_SCODEC_TAS2781"); +/* + * The order of calibrated-data writing is a bit different from the order + * in UEFI. Here is the conversion to match the order of calibrated-data + * writing. + */ +static void cali_cnv(unsigned char *data, unsigned int base, int offset) +{ + __be32 bedata[TASDEV_CALIB_N]; + int i; + + /* r0_reg */ + bedata[0] = cpu_to_be32(*(uint32_t *)&data[base]); + /* r0_low_reg */ + bedata[1] = cpu_to_be32(*(uint32_t *)&data[base + 8]); + /* invr0_reg */ + bedata[2] = cpu_to_be32(*(uint32_t *)&data[base + 4]); + /* pow_reg */ + bedata[3] = cpu_to_be32(*(uint32_t *)&data[base + 12]); + /* tlimit_reg */ + bedata[4] = cpu_to_be32(*(uint32_t *)&data[base + 16]); + + for (i = 0; i < TASDEV_CALIB_N; i++) + memcpy(&data[offset + i * 4 + 1], &bedata[i], + sizeof(bedata[i])); +} + static void tas2781_apply_calib(struct tasdevice_priv *p) { struct calidata *cali_data = &p->cali_data; @@ -86,6 +112,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) for (j = 0, k = 0; j < node_num; j++) { oft = j * 6 + 3; + /* Calibration registers address */ if (tmp_val[oft] == TASDEV_UEFI_CALI_REG_ADDR_FLG) { for (i = 0; i < TASDEV_CALIB_N; i++) { buf = &data[(oft + i + 1) * 4]; @@ -93,6 +120,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) buf[2], buf[3]); } } else { + /* Calibrated data */ l = j * (cali_data->cali_dat_sz_per_dev + 1); if (k >= p->ndev || l > oft * 4) { dev_err(p->dev, "%s: dev sum error\n", @@ -103,8 +131,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) data[l] = k; oft++; - for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i + 1] = data[4 * oft + i]; + cali_cnv(data, 4 * oft, l); k++; } } @@ -127,12 +154,11 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) dev_err(p->dev, "%s: V1 CRC error\n", __func__); return; } - + /* reverse rearrangement in case of overlap */ for (j = p->ndev - 1; j >= 0; j--) { l = j * (cali_data->cali_dat_sz_per_dev + 1); - for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) - data[l + i] = data[p->index * 5 + i]; - data[l+i] = j; + cali_cnv(data, cali_data->cali_dat_sz_per_dev * j, l); + data[l] = j; } } -- 2.43.0

3 4

[PATCH] ASoC: amd: amd_sdw: Add quirks for some new Dell laptops
by Syed Saba Kareem 03 Sep '25

03 Sep '25

Add a quirk to include the codec amplifier function for Dell SKU's listed in quirk table. Note: In these SKU's, the RT722 codec amplifier is excluded, and an external amplifier is used instead. Signed-off-by: Syed Saba Kareem <syed.sabakareem(a)amd.com> --- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/amd/acp/acp-sdw-legacy-mach.c b/sound/soc/amd/acp/acp-sdw-legacy-mach.c index c2197b75a7dd..5a3cfedacbaf 100644 --- a/sound/soc/amd/acp/acp-sdw-legacy-mach.c +++ b/sound/soc/amd/acp/acp-sdw-legacy-mach.c @@ -79,6 +79,22 @@ static const struct dmi_system_id soc_sdw_quirk_table[] = { }, .driver_data = (void *)(ASOC_SDW_CODEC_SPKR), }, + { + .callback = soc_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0DD3"), + }, + .driver_data = (void *)(ASOC_SDW_CODEC_SPKR), + }, + { + .callback = soc_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0DD4"), + }, + .driver_data = (void *)(ASOC_SDW_CODEC_SPKR), + }, {} }; -- 2.43.0

2 1

ucm2: Qualcomm: x1e80100: Add X1E80100-EVK configuration
by GitHub pull_request - opened 03 Sep '25

03 Sep '25

alsa-project/alsa-ucm-conf pull request #608 was opened from ReaLcdcLeQi: The X1E80100-EVK needs basically the same configuration as the X1E80100-CRD. The dts upstream is already in progress. https://lore.kernel.org/all/20250828-hamoa_initial-v8-3-c9d173072a5c@oss.qu… message id <20250828-hamoa_initial-v8-3-c9d173072a5c(a)oss.qualcomm.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/608 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/608.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

A52 producing static noise
by GitHub issues - edited 03 Sep '25

03 Sep '25

alsa-project/alsa-plugins issue #63 was edited from mingww64: Setup is GPU----(HDMI)----TV----(eARC)----Sonos Arc Playing AC3 audio sample works and can confirm the speakers is receiving in AC3 format (Sonos app displaying Dolby Digital 5.1), however using the a52 plugin with `speaker-test` only static noise was coming out intermittently and Sonos app doesn't show any audio format. The GPU is a rx9070xt , I'm unsure if the problem is on the plugin or on AMD's side. ```log Slave: Hardware PCM card 0 'HDA ATI HDMI' device 9 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 3072 period_size : 768 period_time : 16000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 768 period_event : 0 start_threshold : 3072 stop_threshold : 3072 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 0 0 - Front Left 4 - Front Center 1 - Front Right 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 17.888669 0 - Front Left 4 - Front Center 1 - Front Right ^C 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 8.512034 ``` Issue URL : https://github.com/alsa-project/alsa-plugins/issues/63 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

A52 producing static noise
by GitHub issues - opened 03 Sep '25

03 Sep '25

alsa-project/alsa-plugins issue #63 was opened from mingww64: Setup is GPU----(HDMI)----TV----(eARC)----Sonos Arc Playing AC3 audio sample works and can confirm the speakers is receiving in AC3 format (Sonos app displaying Dolby Digital 5.1), however using the a52 plugin with `speaker-test` only static noise was coming out intermittently and Sonos app doesn't show any audio format. ```log Slave: Hardware PCM card 0 'HDA ATI HDMI' device 9 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 3072 period_size : 768 period_time : 16000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 768 period_event : 0 start_threshold : 3072 stop_threshold : 3072 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 0 0 - Front Left 4 - Front Center 1 - Front Right 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 17.888669 0 - Front Left 4 - Front Center 1 - Front Right ^C 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 8.512034 ``` Issue URL : https://github.com/alsa-project/alsa-plugins/issues/63 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[BUG] ALSA FireWire: poll timeout causes JACK to stop (6.10 – 6.17-rc2)
by Илья Кузнецов 03 Sep '25

03 Sep '25

________________________________ Hello ALSA developers, I would like to report a reproducible issue with snd-fireface on Linux kernels 6.10 through 6.17-rc2. ________________________________ System Information Kernel versions tested: 6.10 → 6.17-rc2 Audio interface: RME Fireface 800 Driver in use: snd-fireface Audio server: JACK (tested with jackd and QJackCtl) Distribution: Debian 12 Kernel config: ALSA FireWire drivers enabled, no FFADO in use Chip 1394: Texas Instrumets ________________________________ Steps to Reproduce Boot kernel 6.10 … 6.17-rc2 with snd-fireface loaded. Start JACK with ALSA backend, for example: jackd -d alsa -d hw:Fireface -p 64 -n 3 Use the system normally — during playback, recording, or even when idle. The issue may occur at any time without apparent external cause. Actual Behavior After some idle time, ALSA stops responding and JACK prints: ERROR: ALSA: poll time out, polled for ... usecs ERROR: JackAudioDriver::ProcessAsync: read error, stopping... JACK immediately terminates. The Fireface 800 does not recover until JACK is restarted. ________________________________ Expected Behavior The ALSA FireWire driver should maintain the stream and allow JACK to continue running without manual restart, even when idle. ________________________________ Notes With FFADO backend (jackd -d firewire), the same hardware works reliably and does not exhibit this issue. This suggests that the problem lies in the ALSA FireWire streaming engine rather than the hardware itself. The problem has been reproducible across all tested kernels (vanila, rt, liquorix) from 6.10 through 6.17-rc2. ________________________________ Request Could you please confirm if this issue is known? Is there any ongoing work on the ALSA FireWire streaming engine (snd-fireface) to address these poll timeout conditions? I can provide additional logs (dmesg, journalctl) or test patches if needed. Thank you, Ilya Kuznetsov.

2 2

I2S Stereo ADC Soundcard Hacking
by sylvain.gre＠proton.me 02 Sep '25

02 Sep '25

My idea is to exchange two input signals (CH_L1_IN and CH_L2_IN) at the sampling rate (192kHz) to produce one signal (CH_L_OUT), sampled at 192kHz (small electronic cicuit cf https://electronics.stackexchange.com/questions/754648/i2s-stereo-adc-sound…). In this way, in the digital domain I will receive CH_L1_IN, CH_R1_IN, CH_L2_IN, CH_R2_IN... Of course, the sampling frequency will be decreased by 2 using this method (96kHz instead of 192kHz). But the number of channels is increased by 2 also (4 instead of 2). Where can I begin with alsa to increase the number of channels and adjust the sampling frequency ? Something like a virtual soundcard maybe ? I have skills in dev but don't know where I have to start with alsa fot this hack...

1 0

[PATCH v1] ALSA: hda/tas2781: Fix the order of TAS2781 calibrated-data
by Shenghao Ding 02 Sep '25

02 Sep '25

A bug reported by one of my customers that the order of TAS2781 calibrated-data is incorrect, the correct way is to move R0_Low_%d and insert it between R0_%d and InvR0_%d. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add varialbe csz to store cali_data->cali_dat_sz_per_dev, which is long enough to need two lines. - Add more comments on calibrated-data order --- sound/hda/codecs/side-codecs/tas2781_hda.c | 43 ++++++++++++++++++---- 1 file changed, 35 insertions(+), 8 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index f46d2e06c64f..d5913a727be4 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -33,6 +33,32 @@ const efi_guid_t tasdev_fct_efi_guid[] = { }; EXPORT_SYMBOL_NS_GPL(tasdev_fct_efi_guid, "SND_HDA_SCODEC_TAS2781"); +/* + * The order of calibrated-data writing is a bit different from the order + * in UEFI. Here is the conversion to match the order of calibrated-data + * writing. + */ +static void cali_cnv(unsigned char *data, unsigned int base, int offset) +{ + __be32 bedata[TASDEV_CALIB_N]; + int i; + + /* r0_reg */ + bedata[0] = cpu_to_be32(*(uint32_t *)&data[base]); + /* r0_low_reg */ + bedata[1] = cpu_to_be32(*(uint32_t *)&data[base + 8]); + /* invr0_reg */ + bedata[2] = cpu_to_be32(*(uint32_t *)&data[base + 4]); + /* pow_reg */ + bedata[3] = cpu_to_be32(*(uint32_t *)&data[base + 12]); + /* tlimit_reg */ + bedata[4] = cpu_to_be32(*(uint32_t *)&data[base + 16]); + + for (i = 0; i < TASDEV_CALIB_N; i++) + memcpy(&data[offset + i * 4 + 1], &bedata[i], + sizeof(bedata[i])); +} + static void tas2781_apply_calib(struct tasdevice_priv *p) { struct calidata *cali_data = &p->cali_data; @@ -46,6 +72,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) TASDEVICE_REG(0, 0x13, 0x70), TASDEVICE_REG(0, 0x18, 0x7c), }; + unsigned int csz = cali_data->cali_dat_sz_per_dev; unsigned int crc, oft, node_num; unsigned char *buf; int i, j, k, l; @@ -86,6 +113,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) for (j = 0, k = 0; j < node_num; j++) { oft = j * 6 + 3; + /* Calibration registers address */ if (tmp_val[oft] == TASDEV_UEFI_CALI_REG_ADDR_FLG) { for (i = 0; i < TASDEV_CALIB_N; i++) { buf = &data[(oft + i + 1) * 4]; @@ -93,7 +121,8 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) buf[2], buf[3]); } } else { - l = j * (cali_data->cali_dat_sz_per_dev + 1); + /* Calibrated data */ + l = j * (csz + 1); if (k >= p->ndev || l > oft * 4) { dev_err(p->dev, "%s: dev sum error\n", __func__); @@ -103,8 +132,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) data[l] = k; oft++; - for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i + 1] = data[4 * oft + i]; + cali_cnv(data, 4 * oft, l); k++; } } @@ -127,12 +155,11 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) dev_err(p->dev, "%s: V1 CRC error\n", __func__); return; } - + /* reverse rearrangement in case of overlap */ for (j = p->ndev - 1; j >= 0; j--) { - l = j * (cali_data->cali_dat_sz_per_dev + 1); - for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) - data[l + i] = data[p->index * 5 + i]; - data[l+i] = j; + l = j * (csz + 1); + cali_cnv(data, csz * j, l); + data[l] = j; } } -- 2.43.0

2 1

tlv: ignore xdir when type is DB_MINMAX_MUTE
by GitHub pull_request - opened 02 Sep '25

02 Sep '25

alsa-project/alsa-lib pull request #474 was opened from qaqland: Muting caused by rounding is meaningless, and it is also inconvenient for upper-layer applications to handling muting issues caused by DB_MINMAX_MUTE or xdir. Thus, the restriction on xdir is removed. Request URL : https://github.com/alsa-project/alsa-lib/pull/474 Patch URL : https://github.com/alsa-project/alsa-lib/pull/474.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ALSA: hda/tas2781: Fix the order of TAS2563 calibrated-data
by Shenghao Ding 01 Sep '25

01 Sep '25

A bug reported by one of my customers that the order of TAS2563 calibrated-data is incorrect, the correct way is to move R0_Low_%d and insert it between R0_%d and InvR0_%d. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index dbd71e173119..853ff5d87966 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -313,7 +313,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h) { efi_guid_t efi_guid = tasdev_fct_efi_guid[LENOVO]; char *vars[TASDEV_CALIB_N] = { - "R0_%d", "InvR0_%d", "R0_Low_%d", "Power_%d", "TLim_%d" + "R0_%d", "R0_Low_%d", "InvR0_%d", "Power_%d", "TLim_%d" }; efi_char16_t efi_name[TAS2563_CAL_VAR_NAME_MAX]; unsigned long max_size = TAS2563_CAL_DATA_SIZE; -- 2.43.0

3 2

[PATCH 1/2] ALSA: hda: tas2781: fix tas2563 EFI data endianness
by Gergo Koteles 30 Aug '25

30 Aug '25

Before conversion to unify the calibration data management, the tas2563_apply_calib() function performed the big endian conversion and wrote the calibration data to the device. The writing is now done by the common tasdev_load_calibrated_data() function, but without conversion. Put the values into the calibration data buffer with the expected endianness. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Cc: <stable(a)vger.kernel.org> Signed-off-by: Gergo Koteles <soyer(a)irl.hu> --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index e34b17f0c9b9..1eac751ab2a6 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -310,6 +310,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h) struct cali_reg *r = &cd->cali_reg_array; unsigned int offset = 0; unsigned char *data; + __be32 bedata; efi_status_t status; unsigned int attr; int ret, i, j, k; @@ -351,6 +352,8 @@ static int tas2563_save_calibration(struct tas2781_hda *h) i, j, status); return -EINVAL; } + bedata = cpu_to_be32(*(uint32_t *)&data[offset]); + memcpy(&data[offset], &bedata, sizeof(bedata)); offset += TAS2563_CAL_DATA_SIZE; } } -- 2.51.0

2 2

[BUG] ALSA FireWire: poll timeout causes JACK to stop (6.10 – 6.17-rc2)
by Илья Кузнецов 30 Aug '25

30 Aug '25

------------------------------ Hello ALSA developers, I would like to report a reproducible issue with *snd-fireface* on Linux kernels 6.10 through 6.17-rc2. ------------------------------ System Information - *Kernel versions tested:* 6.10 → 6.17-rc2 - *Audio interface:* RME Fireface 800 - *Driver in use:* snd-fireface - *Audio server:* JACK (tested with jackd and QJackCtl) - *Distribution:* Debian 12 - *Kernel config:* ALSA FireWire drivers enabled, no FFADO in use - *Chip 1394: *Texas Instrumets ------------------------------ Steps to Reproduce 1. Boot kernel 6.10 … 6.17-rc2 with snd-fireface loaded. 2. Start JACK with ALSA backend, for example: jackd -d alsa -d hw:Fireface -p 64 -n 3 3. Use the system normally — during playback, recording, or even when idle. The issue may occur at any time without apparent external cause. Actual Behavior After some idle time, ALSA stops responding and JACK prints: ERROR: ALSA: poll time out, polled for ... usecs ERROR: JackAudioDriver::ProcessAsync: read error, stopping... JACK immediately terminates. The Fireface 800 does not recover until JACK is restarted. ------------------------------ Expected Behavior The ALSA FireWire driver should maintain the stream and allow JACK to continue running without manual restart, even when idle. ------------------------------ Notes 4. With *FFADO* backend (jackd -d firewire), the same hardware works reliably and does not exhibit this issue. 5. This suggests that the problem lies in the ALSA FireWire streaming engine rather than the hardware itself. 6. The problem has been reproducible across all tested kernels (vanila, rt, liquorix) from *6.10 through 6.17-rc2*. ------------------------------ Request Could you please confirm if this issue is known? Is there any ongoing work on the ALSA FireWire streaming engine ( snd-fireface) to address these poll timeout conditions? I can provide additional logs (dmesg, journalctl) or test patches if needed. Thank you, Ilya Kuznetsov.

1 0

URGENT: ALSAfirewire Drivers Introduce 90ms delay - Linux is Unusable for Pro Audio
by M. Armsby 29 Aug '25

29 Aug '25

Dear maintainers, Linux is being ridiculed in the professional audio community — not because it can't do low-latency audio, but because the ALSA FireWire stack actively breaks it. Professional gear costing thousands of dollars like Motu RME Focusrite etc. etc. Since kernel 5.16+, FireWire audio devices are correctly detected and functional — but they are burdened with an artificial +90ms latency offset due to missing timing metadata (e.g., cycle-match delay, transmission latency) in the driver's interface to userspace. PipeWire and WirePlumber rely on accurate latency reporting. When it's missing, they fall back to worst-case assumptions — adding ~90ms of delay. This is NOT a PipeWire bug. It is a FAILURE of the ALSA FireWire drivers to expose timing data that the hardware and firmware already provide. This bug: - Renders high-end interfaces (RME, Focusrite, Tascam, Motu, Terratec) unusable for real-time monitoring - Forces users to hack WirePlumber with hardcoded quantum rules - Damages Linux's credibility in music production - Makes Linux a JOKE in studios worldwide There is NO excuse for this in 2025. We have workarounds (see: https://forum.cockos.com/showthread.php?t=274978), but they are fragile, device-specific, and do not scale. We demand: 1. Immediate exposure of accurate device latency via ALSA control or hwdep interface 2. Proper reporting of transmission/cycle delay to userspace 3. A timeline for fixing this in mainline This is not a "nice-to-have". It is a **critical defect** in a core multimedia subsystem now using pipewire as standard. Linux can and must do better. Many are waiting to move to Linux, finally leaving windows. — Martin Armsby (Martin A - maa) Professional Audio Producer / Linux Advocate Please see: Imagine if Linux Audio just worked - Firewire <https://forum.cockos.com/showthread.php?t=302305> and <https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4785> Thank you for reading. -- Martin Armsby -- Martin Armsby

1 1

[GIT PULL] ASoC fixes for v6.17-rc3
by Mark Brown 29 Aug '25

29 Aug '25

The following changes since commit 1b237f190eb3d36f52dffe07a40b5eb210280e00: Linux 6.17-rc3 (2025-08-24 12:04:12 -0400) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-rc3 for you to fetch changes up to 3e7fd1febc3156d3d98fba229399a13b12d69707: ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op (2025-08-28 21:15:49 +0200) ---------------------------------------------------------------- ASoC: Fixes for v6.17 The main fixes here are for some of the cleanups done in the core in this release, we had broken component lookup in the case with a single bus and DMA controller. Otherwise it's driver specific changes, the shortlogs for the Intel WCL and rsnd drivers look like minor cleanups but are actually bugfixes (adding an op needed for correct functionality and reverting an inappropriate helper usage). ---------------------------------------------------------------- Ajye Huang (1): ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op Cryolitia PukNgae (1): ASoC: codecs: idt821034: fix wrong log in idt821034_chip_direction_output() Kuninori Morimoto (3): ASoC: soc-core: care NULL dirver name on snd_soc_lookup_component_nolocked() ASoC: soc-core: tidyup snd_soc_lookup_component_nolocked() ASoC: rsnd: tidyup direction name on rsnd_dai_connect() Mark Brown (1): ASoC: fixup snd_soc_lookup_component_nolocked() sound/soc/codecs/idt821034.c | 2 +- sound/soc/renesas/rcar/core.c | 2 +- sound/soc/soc-core.c | 25 +++++++++++++++---------- sound/soc/sof/intel/ptl.c | 1 + 4 files changed, 18 insertions(+), 12 deletions(-)

2 1

[PATCH v3 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 29 Aug '25

29 Aug '25

Add tas2020, tas2118, tas2120, tas2320, tas5825 support in tas2781 driver. Tas2118, tas2x20 have no on-chip DSP, tas5825 have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v3: - No update. v2: - Update the mail list for maintainers of yaml file --- include/sound/tas2781.h | 11 +- include/sound/tas2x20-tlv.h | 259 ++++++++++++++++++++++++++ sound/soc/codecs/tas2781-comlib-i2c.c | 2 + sound/soc/codecs/tas2781-i2c.c | 156 +++++++++++----- 4 files changed, 381 insertions(+), 47 deletions(-) create mode 100644 include/sound/tas2x20-tlv.h diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index f0aefc04a957..73ac447b9327 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -51,7 +51,9 @@ /* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) + +#define TAS5825_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) @@ -110,8 +112,14 @@ #define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { + TAS2020, + TAS2118, + TAS2120, + TAS2320, TAS2563, TAS2781, + TAS5825, + TAS_OTHERS, }; enum dspbin_type { @@ -194,6 +202,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const unsigned char (*dvc_tlv_table)[4]; const char *name_prefix; unsigned char ndev; unsigned int dspbin_typ; diff --git a/include/sound/tas2x20-tlv.h b/include/sound/tas2x20-tlv.h new file mode 100644 index 000000000000..6e6bcec4a0a1 --- /dev/null +++ b/include/sound/tas2x20-tlv.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2x20/TAS2118 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2x20/TAS2118 hda driver implements for one, two, or even multiple +// TAS2x20/TAS2118 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2X20_TLV_H__ +#define __TAS2X20_TLV_H__ + +#define TAS2X20_DVC_LEVEL TASDEVICE_REG(0x0, 0x2, 0x0c) +#define TAS2X20_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x07) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_amp_tlv, 2100, 50, 0); + +/* pow(10, db/20) * pow(2,22) */ +static const __maybe_unused unsigned char tas2x20_dvc_table[][4] = { + { 0X00, 0X00, 0X0D, 0X00 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0X00 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0X00 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0X00 }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0X00 }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0X00 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X00 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X00 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X00 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0X00 }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X00 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X00 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0X00 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X00 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X00 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X00 }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X00 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X00 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0X00 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X00 }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X00 }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0X00 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0X00 }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0X00 }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X00 }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0X00 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X00 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X00 }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X00 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X00 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0X00 }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0X00 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0X00 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X00 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X00 }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X00 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X00 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X00 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0X00 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X00 }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0X00 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0X00 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0X00 }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0X00 }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X00 }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X00 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X00 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0X00 }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0X00 }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0X00 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0X00 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X00 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0X00 }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X00 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X00 }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0X00 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0X00 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0X00 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X00 }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X00 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X00 }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X00 }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X00 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X00 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X00 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X00 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0X00 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X00 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0X00 }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X00 }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0X00 }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X00 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0X00 }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0X00 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X00 }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0X00 }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X00 }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X00 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X00 }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0X00 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X00 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X00 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0X00 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0X00 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X00 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X00 }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X00 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0X00 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0X00 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X00 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X00 }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X00 }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X00 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X00 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0X00 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X00 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X00 }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0X00 }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X00 }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0X00 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X00 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0X00 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X00 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X00 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X00 }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0X00 }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X00 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X00 }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0X00 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X00 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0X00 }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X00 }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0X00 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X00 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X00 }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0X00 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X00 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X00 }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X00 }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X00 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0X00 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0X00 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0X00 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X00 }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X00 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X00 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X00 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0X00 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X00 }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0X00 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X00 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X00 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X00 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0X00 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X00 }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0X00 }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0X00 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0X00 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X00 }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X00 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0X00 }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X00 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X00 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0X00 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X00 }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0X00 }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X00 }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X00 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X00 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0X00 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X00 }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X00 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0X00 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X00 }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X00 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0X00 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X00 }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X00 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X00 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0X00 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X00 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0X00 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X00 }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X00 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0X00 }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X00 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X00 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X00 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X00 }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0X00 }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0X00 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0X00 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0X00 }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X00 }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X00 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0X00 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0X00 }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0X00 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X00 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X00 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0X00 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0X00 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X00 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X00 }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X00 }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0X00 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X00 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X00 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X00 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0X00 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0X00 }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X00 }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0X00 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X00 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X00 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0X00 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X00 }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X00 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X00 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0X00 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X00 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X00 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X00 }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X00 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0X00 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X00 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X00 }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0X00 }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X00 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X00 }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0X00 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X00 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0X00 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X00 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X00 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X00 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X00 }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X00 }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X00 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X00 }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0X00 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0X00 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X00 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0X00 }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X00 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0X00 }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X00 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X00 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0X00 }, /* 5.5db */ + { 0X7F, 0XB2, 0X61, 0X00 }, /* 6.0db */ +}; +#endif diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c index c078bb0a8437..764304604e2c 100644 --- a/sound/soc/codecs/tas2781-comlib-i2c.c +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -320,6 +320,8 @@ void tasdevice_reset(struct tasdevice_priv *tas_dev) for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, TASDEVICE_REG_SWRESET, + tas_dev->chip_id == TAS5825 ? + TAS5825_REG_SWRESET_RESET : TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index ea3cdb8553de..cf6bd4d63f31 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,8 +30,10 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2x20-tlv.h> #include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include <linux/unaligned.h> #define X2563_CL_STT_VAL(xreg, xval) \ @@ -98,16 +100,26 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { }; static const struct i2c_device_id tasdevice_id[] = { + { "tas2020", TAS2020 }, + { "tas2118", TAS2118 }, + { "tas2120", TAS2120 }, + { "tas2320", TAS2320 }, { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, + { "tas5825", TAS5825 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); #ifdef CONFIG_OF static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2020" }, + { .compatible = "ti,tas2118" }, + { .compatible = "ti,tas2120" }, + { .compatible = "ti,tas2320" }, { .compatible = "ti,tas2563" }, { .compatible = "ti,tas2781" }, + { .compatible = "ti,tas5825" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -797,7 +809,7 @@ static int tasdev_nop_get( return 0; } -static int tas2563_digital_gain_get( +static int tasdevice_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -823,15 +835,15 @@ static int tas2563_digital_gain_get( while (r > 1 + l) { mid = (l + r) / 2; - ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + ar_mid = get_unaligned_be32(tas_dev->dvc_tlv_table[mid]); if (target < ar_mid) r = mid; else l = mid; } - ar_l = get_unaligned_be32(tas2563_dvc_table[l]); - ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + ar_l = get_unaligned_be32(tas_dev->dvc_tlv_table[l]); + ar_r = get_unaligned_be32(tas_dev->dvc_tlv_table[r]); /* find out the member same as or closer to the current volume */ ucontrol->value.integer.value[0] = @@ -841,7 +853,7 @@ static int tas2563_digital_gain_get( return 0; } -static int tas2563_digital_gain_put( +static int tasdevice_digital_gain_put( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -867,7 +879,7 @@ static int tas2563_digital_gain_put( } volrd = get_unaligned_be32(&data[0]); - volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + volwr = get_unaligned_be32(tas_dev->dvc_tlv_table[vol]); if (volrd == volwr) { rc = 0; @@ -876,7 +888,7 @@ static int tas2563_digital_gain_put( for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_bulk_write(tas_dev, i, reg, - (unsigned char *)tas2563_dvc_table[vol], 4); + (unsigned char *)tas_dev->dvc_tlv_table[vol], 4); if (ret) { dev_err(tas_dev->dev, "%s, set digital vol error in dev %d\n", @@ -892,11 +904,6 @@ static int tas2563_digital_gain_put( return rc; } -static const struct snd_kcontrol_new tasdevice_snd_controls[] = { - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tasdev_force_fwload_get, tasdev_force_fwload_put), -}; - static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, tasdev_nop_get, tasdev_calib_stop_put), @@ -907,6 +914,16 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), }; +static const struct snd_kcontrol_new tas2x20_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2X20_AMP_LEVEL, + 0, 0, 42, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas2x20_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2X20_DVC_LEVEL, + 0, 0, ARRAY_SIZE(tas2x20_dvc_table) - 1, 0, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, + tas2x20_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -916,6 +933,15 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, tas2781_dvc_tlv), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_cali_controls[] = { SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), }; @@ -923,7 +949,7 @@ static const struct snd_kcontrol_new tas2781_cali_controls[] = { static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, - tas2563_digital_gain_get, tas2563_digital_gain_put, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, tas2563_dvc_tlv), }; @@ -968,7 +994,7 @@ static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; - uinfo->value.integer.min = TAS2563; + uinfo->value.integer.min = TAS2020; uinfo->value.integer.max = TAS2781; return 0; @@ -1168,9 +1194,9 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *active_dev_num, *chip_id; + char *active_dev_num, *chip_id, *fw_load; char *conf_name, *prog_name; - int nr_controls = 4; + int nr_controls = 5; int mix_index = 0; /* Alloc kcontrol via devm_kzalloc, which don't manually @@ -1228,6 +1254,19 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; + fw_load = devm_kstrdup(tas_priv->dev, "Speaker Force Firmware Load", + GFP_KERNEL); + if (!fw_load) + return -ENOMEM; + + dsp_ctrls[mix_index].name = fw_load; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_info_bool_ext; + dsp_ctrls[mix_index].put = tasdev_force_fwload_put; + dsp_ctrls[mix_index].get = tasdev_force_fwload_get; + dsp_ctrls[mix_index].private_value = 0UL; + mix_index++; + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); } @@ -1587,6 +1626,9 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + /* There is no DSP firmware required for TAS2118/2X20. */ + if (tas_priv->chip_id < TAS2563) + goto out; if (tas_priv->name_prefix) scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); @@ -1608,34 +1650,37 @@ static void tasdevice_fw_ready(const struct firmware *fmw, dev_err(tas_priv->dev, "dsp controls error\n"); goto out; } - - ret = tasdevice_create_cali_ctrls(tas_priv); - if (ret) { - dev_err(tas_priv->dev, "cali controls error\n"); - goto out; - } - tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - /* If calibrated data occurs error, dsp will still works with default - * calibrated data inside algo. - */ - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->name_prefix) - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, - tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - else - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s_cal_0x%02x.bin", tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - ret = tas2781_load_calibration(tas_priv, - tas_priv->cal_binaryname[i], i); - if (ret != 0) - dev_err(tas_priv->dev, - "%s: load %s error, default will effect\n", - __func__, tas_priv->cal_binaryname[i]); + /* There is no calibration required for TAS5825. */ + if (tas_priv->chip_id < TAS5825) { + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + /* If calibrated data occurs error, dsp will still works + * with default calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", + tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, keep default.\n", + __func__, tas_priv->cal_binaryname[i]); + } } tasdevice_prmg_load(tas_priv, 0); @@ -1658,7 +1703,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, comp, &acoustic_ctl_fops); #endif out: - if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + if ((tas_priv->fw_state == TASDEVICE_RCA_FW_OK) && + (tas_priv->chip_id >= TAS2563)) { /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } @@ -1804,13 +1850,27 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) int rc; switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + p = (struct snd_kcontrol_new *)tas2x20_snd_controls; + size = ARRAY_SIZE(tas2x20_snd_controls); + tas_priv->dvc_tlv_table = tas2x20_dvc_table; + break; case TAS2781: p = (struct snd_kcontrol_new *)tas2781_snd_controls; size = ARRAY_SIZE(tas2781_snd_controls); break; + case TAS5825: + p = (struct snd_kcontrol_new *)tas5825_snd_controls; + size = ARRAY_SIZE(tas5825_snd_controls); + break; default: p = (struct snd_kcontrol_new *)tas2563_snd_controls; size = ARRAY_SIZE(tas2563_snd_controls); + tas_priv->dvc_tlv_table = tas2563_dvc_table; + break; } rc = snd_soc_add_component_controls(codec, p, size); @@ -1850,8 +1910,6 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tasdevice_snd_controls, - .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, @@ -1967,7 +2025,13 @@ static void tasdevice_i2c_remove(struct i2c_client *client) #ifdef CONFIG_ACPI static const struct acpi_device_id tasdevice_acpi_match[] = { - { "TAS2781", TAS2781 }, + { "TXNW2020", TAS2020 }, + { "TXNW2118", TAS2118 }, + { "TXNW2120", TAS2120 }, + { "TXNW2320", TAS2320 }, + { "TXNW2563", TAS2563 }, + { "TXNW2781", TAS2781 }, + { "TXNW5825", TAS5825 }, {}, }; -- 2.43.0

1 1

when i click apply HDAjackretask is not respoding
by GitHub issues - opened 29 Aug '25

29 Aug '25

alsa-project/alsa-tools issue #40 was opened from m33rxj: I'm using a hp envy x360 and when i try change and override the pins to allow the front speakers and click apply, it says on the gui 'its not responding' it happened to me on ubuntu and now on cachyos. I'm not sure if my codec is unsupported? its RealTek ALC245 I'm trying to work on but even on the intel codec I get the same error. I forgot to mention I have pipewire but I'm unsure what I used for ubuntu at the time Any chance I can get some help incase I'm doing something wrong. <img width="656" height="869" alt="Image" src="https://github.com/user-attachments/assets/dfdd4d42-3979-4e89-9234-b49726d2…" /> <img width="808" height="295" alt="Image" src="https://github.com/user-attachments/assets/42f65770-e19d-40d3-8ec7-0521e345…" /> Issue URL : https://github.com/alsa-project/alsa-tools/issues/40 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH v3] ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op
by Ajye Huang 28 Aug '25

28 Aug '25

Add the missing op in the device description to avoid issues with jack detection. Fixes: 6b04629ae97a ("ASoC: SOF: Intel: add initial support for WCL") Acked-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- v2: Add Fixes: 6b04629ae97a ("ASoC: SOF: Intel: add initial support for WCL") v3: Add Acked-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> sound/soc/sof/intel/ptl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/ptl.c b/sound/soc/sof/intel/ptl.c index 1bc1f54c470d..4633cd01e7dd 100644 --- a/sound/soc/sof/intel/ptl.c +++ b/sound/soc/sof/intel/ptl.c @@ -143,6 +143,7 @@ const struct sof_intel_dsp_desc wcl_chip_info = { .read_sdw_lcount = hda_sdw_check_lcount_ext, .check_sdw_irq = lnl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, -- 2.25.1

2 1

ucm2: MediaTek: mt8391-evk: Add alsa-ucm support
by GitHub pull_request - opened 28 Aug '25

28 Aug '25

alsa-project/alsa-ucm-conf pull request #607 was opened from zoran055: Add alsa-ucm support for the MediaTek mt8391-evk platform. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/607 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/607.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ALSA: hda/tas2781: Fix EFI name for calibration beginning with 1 instead of 0
by Shenghao Ding 27 Aug '25

27 Aug '25

A bug reported by one of my customers that EFI name beginning with 0 instead of 1. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - remove unrelated change v1: - Fix EFI name beginning with 1 instead of 0 - Add extra comments on EFI name for calibration - Remove an extra space --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index ed7771ab9475..635cbd8820ac 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -340,7 +340,8 @@ static int tas2563_save_calibration(struct tas2781_hda *h) data[offset] = i; offset++; for (j = 0; j < TASDEV_CALIB_N; ++j) { - ret = snprintf(var8, sizeof(var8), vars[j], i); + /* EFI name for calibration started with 1, not 0 */ + ret = snprintf(var8, sizeof(var8), vars[j], i + 1); if (ret < 0 || ret >= sizeof(var8) - 1) { dev_err(p->dev, "%s: Read %s failed\n", @@ -349,7 +350,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h) } /* * Our variable names are ASCII by construction, but - * EFI names are wide chars. Convert and zero-pad. + * EFI names are wide chars. Convert and zero-pad. */ memset(efi_name, 0, sizeof(efi_name)); for (k = 0; k < sizeof(var8) && var8[k]; k++) -- 2.43.0

4 5

Add UCM for RME Fireface UCX
by GitHub pull_request - opened 27 Aug '25

27 Aug '25

alsa-project/alsa-ucm-conf pull request #606 was opened from vascosantos: Adds UCM for RME Fireface UCX (heavily based on RME Fireface UCX II config). Device is assumed in CC mode. #605 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/606 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/606.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Feature request: Add support for RME Fireface UCX
by GitHub issues - opened 27 Aug '25

27 Aug '25

alsa-project/alsa-ucm-conf issue #605 was opened from vascosantos: I see there is already support for the RME Fireface UCX II. Could you please add support for the previous version, RME Fireface UCX? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/605 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3] ALSA: hda/tas2781: Fix EFI name for calibration beginning with 1 instead of 0
by Shenghao Ding 27 Aug '25

27 Aug '25

A bug reported by one of my customers that EFI name beginning with 0 instead of 1. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - remove unrelated change v1: - Fix EFI name beginning with 1 instead of 0 - Add extra comments on EFI name for calibration --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index ed7771ab9475..dbd71e173119 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -340,8 +340,8 @@ static int tas2563_save_calibration(struct tas2781_hda *h) data[offset] = i; offset++; for (j = 0; j < TASDEV_CALIB_N; ++j) { - ret = snprintf(var8, sizeof(var8), vars[j], i); - + /* EFI name for calibration started with 1, not 0 */ + ret = snprintf(var8, sizeof(var8), vars[j], i + 1); if (ret < 0 || ret >= sizeof(var8) - 1) { dev_err(p->dev, "%s: Read %s failed\n", __func__, var8); -- 2.43.0

2 1

ADC Soundcard hack : Increase by 2 the number of channels for any adc soundcard
by Sylvain Grenier 27 Aug '25

27 Aug '25

Hello alsa team, To begin, thank you all the team for what you do, the alsa-project is a wonderfull open-source project! My goal is to increase the number of channels of my adc soundcard by separate once again a left-right for one channel. Using a small electronic scheme like this (it's a first simulation try, I have to test and improve), we can multiply the number of channels by 2. [image.png] Of course, the sampling frequency will be decreased by 2 using this method. Where can I begin with alsa to increase the number of channels and adjust the sampling frequency ? Something like a virtual soundcard maybe ? I have skills in dev but don't know where I have to start with alsa fot this hack... Best regards, Sylvain Grenier

1 0

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - edited 26 Aug '25

26 Aug '25

alsa-project/alsa-lib issue #472 was edited from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from multiple soundcards on the Raspberry Pi. The results are a garbled mess. ==== EDIT 26/08 ==== Having probed the issue some more, I would like to add more detail: The devices and drivers I try to use are: ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Device_1 [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` ...Curiously, I have actually had success using the multi plugin to record using a pro audio interface AND ONE `USB PnP Sound Device`, in this set up ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: USB [Scarlett 2i2 USB], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` HOWEVER, if I try to add another `USB PnP Sound Device` to the multi setup, then the result is a garbled mess. My conclusion is that the multi plugin works generally, but it does NOT work for two or more `USB PnP Sound Device`. Something tells me that the plugin cannot differentiate the addresses of the audio frames held in two or more of these cheap USB microphones. I have a more verbose post of my issue here: https://www.reddit.com/r/linuxaudio/comments/1m24po8/recording_on_multiple_… ===EDIT 26/08 The following diagnosis was wrong: I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - edited 26 Aug '25

26 Aug '25

alsa-project/alsa-lib issue #472 was edited from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from multiple soundcards on the Raspberry Pi. The results are a garbled mess. ==== EDIT 26/08 ==== Having probed the issue some more, I would like to add more detail: The devices and drivers I try to use are: ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Device_1 [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` ...Curiously, I have actually had success using the multi plugin to record using a pro audio interface AND ONE `USB PnP Sound Device`, in this set up ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: USB [Scarlett 2i2 USB], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0``` HOWEVER, if I try to add another `USB PnP Sound Device` to the multi setup, then the result is a garbled mess. My conclusion is that the multi plugin works generally, but it does NOT work for two or more `USB PnP Sound Device`. Something tells me that the plugin cannot differentiate the addresses of the audio frames held in two or more of these cheap USB microphones. I have a more verbose post of my issue here: https://www.reddit.com/r/linuxaudio/comments/1m24po8/recording_on_multiple_… ===EDIT 26/08 The following diagnosis was wrong: I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - edited 26 Aug '25

26 Aug '25

alsa-project/alsa-lib issue #472 was edited from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from multiple soundcards on the Raspberry Pi. The results are a garbled mess. ==== EDIT 26/08 ==== Having probed the issue some more, I would like to add more detail: The devices and drivers I try to use are: `**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Device_1 [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ` ...Curiously, I have actually had success using the multi plugin to record using a pro audio interface AND ONE `USB PnP Sound Device`, in this set up `**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: USB [Scarlett 2i2 USB], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0` HOWEVER, if I try to add another `USB PnP Sound Device` to the multi setup, then the result is a garbled mess. My conclusion is that the multi plugin works generally, but it does NOT work for two or more `USB PnP Sound Device`. Something tells me that the plugin cannot differentiate the addresses of the audio frames held in two or more of these cheap USB microphones. I have a more verbose post of my issue here: https://www.reddit.com/r/linuxaudio/comments/1m24po8/recording_on_multiple_… ===EDIT 26/08 The following diagnosis was wrong: I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op
by Ajye Huang 26 Aug '25

26 Aug '25

Add the missing op in the device description to avoid issues with jack detection. Fixes: 6b04629ae97a ("ASoC: SOF: Intel: add initial support for WCL") Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- v2: Add Fixes: 6b04629ae97a ("ASoC: SOF: Intel: add initial support for WCL") sound/soc/sof/intel/ptl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/ptl.c b/sound/soc/sof/intel/ptl.c index 1bc1f54c470d..4633cd01e7dd 100644 --- a/sound/soc/sof/intel/ptl.c +++ b/sound/soc/sof/intel/ptl.c @@ -143,6 +143,7 @@ const struct sof_intel_dsp_desc wcl_chip_info = { .read_sdw_lcount = hda_sdw_check_lcount_ext, .check_sdw_irq = lnl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, -- 2.25.1

1 1

[PATCH v1] ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op
by Ajye Huang 26 Aug '25

26 Aug '25

Add the missing op in the device description to avoid issues with jack detection. Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- sound/soc/sof/intel/ptl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/ptl.c b/sound/soc/sof/intel/ptl.c index 1bc1f54c470d..4633cd01e7dd 100644 --- a/sound/soc/sof/intel/ptl.c +++ b/sound/soc/sof/intel/ptl.c @@ -143,6 +143,7 @@ const struct sof_intel_dsp_desc wcl_chip_info = { .read_sdw_lcount = hda_sdw_check_lcount_ext, .check_sdw_irq = lnl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, -- 2.25.1

2 2

[PATCH v1] ALSA: hda/tas2781: Fix EFI name for calibration beginning with 1 instead of 0
by Shenghao Ding 26 Aug '25

26 Aug '25

A bug reported by one of my customers that EFI name beginning with 0 instead of 1, and code clean for the string checking. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Fix EFI name beginning with 1 instead of 0 - Code clean for the string checking - Add extra comments on EFI name for calibration - Remove an extra space --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index ed7771ab9475..fecd5eac739b 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -340,16 +340,17 @@ static int tas2563_save_calibration(struct tas2781_hda *h) data[offset] = i; offset++; for (j = 0; j < TASDEV_CALIB_N; ++j) { - ret = snprintf(var8, sizeof(var8), vars[j], i); + /* EFI name for calibration started with 1, not 0 */ + ret = snprintf(var8, sizeof(var8), vars[j], i + 1); - if (ret < 0 || ret >= sizeof(var8) - 1) { + if (ret != strlen(var8)) { dev_err(p->dev, "%s: Read %s failed\n", __func__, var8); return -EINVAL; } /* * Our variable names are ASCII by construction, but - * EFI names are wide chars. Convert and zero-pad. + * EFI names are wide chars. Convert and zero-pad. */ memset(efi_name, 0, sizeof(efi_name)); for (k = 0; k < sizeof(var8) && var8[k]; k++) -- 2.43.0

2 1

my duo cast volume is too low
by GitHub issues - opened 25 Aug '25

25 Aug '25

alsa-project/alsa-ucm-conf issue #604 was opened from reflexionn: hello i bought the hyperx duocast mic for recording reasons i have a small youtube channel that goes by the name vironio i was dissapointed with the mic performance because it has very low volume and if i wanna record i have to be uncomfortably close to the microphone i thought thats just how the mic is and i lived with it i set the gain to around 70% because anything more would lead to weird noise that comes from the mic itself and i just record and then put the recording in adobe audition add a lot of effects and noise cancelation and audio gate and then i amplify the audio by around 9db so its hearable but not so long ago i saw a tiktoker with the same mic but he has a much higher volume and can use the microphone from around 2 to 3 inches away with the quality being really good so i thought its not actually the mics fault and i just didnt set it up right can anyone help me? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/604 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: MediaTek: mt8395-evk: Add support for SOF
by GitHub pull_request - edited 25 Aug '25

25 Aug '25

alsa-project/alsa-ucm-conf pull request #549 was edited from julien-massot: Add support for sof-mt8395-evk. Currently, enable only the minimum: earphone output and HDMI output. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/549 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/549.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 22 Aug '25

22 Aug '25

Add tas2020, tas2118, tas2120, tas2320, tas5825 support in tas2781 driver. Tas2118, tas2x20 have no on-chip DSP, tas5825 have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2: - Update the mail list for maintainers of yaml file --- include/sound/tas2781.h | 11 +- include/sound/tas2x20-tlv.h | 259 ++++++++++++++++++++++++++ sound/soc/codecs/tas2781-comlib-i2c.c | 2 + sound/soc/codecs/tas2781-i2c.c | 156 +++++++++++----- 4 files changed, 381 insertions(+), 47 deletions(-) create mode 100644 include/sound/tas2x20-tlv.h diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index f0aefc04a957..73ac447b9327 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -51,7 +51,9 @@ /* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) + +#define TAS5825_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) @@ -110,8 +112,14 @@ #define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { + TAS2020, + TAS2118, + TAS2120, + TAS2320, TAS2563, TAS2781, + TAS5825, + TAS_OTHERS, }; enum dspbin_type { @@ -194,6 +202,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const unsigned char (*dvc_tlv_table)[4]; const char *name_prefix; unsigned char ndev; unsigned int dspbin_typ; diff --git a/include/sound/tas2x20-tlv.h b/include/sound/tas2x20-tlv.h new file mode 100644 index 000000000000..6e6bcec4a0a1 --- /dev/null +++ b/include/sound/tas2x20-tlv.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2x20/TAS2118 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2x20/TAS2118 hda driver implements for one, two, or even multiple +// TAS2x20/TAS2118 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2X20_TLV_H__ +#define __TAS2X20_TLV_H__ + +#define TAS2X20_DVC_LEVEL TASDEVICE_REG(0x0, 0x2, 0x0c) +#define TAS2X20_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x07) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_amp_tlv, 2100, 50, 0); + +/* pow(10, db/20) * pow(2,22) */ +static const __maybe_unused unsigned char tas2x20_dvc_table[][4] = { + { 0X00, 0X00, 0X0D, 0X00 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0X00 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0X00 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0X00 }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0X00 }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0X00 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X00 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X00 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X00 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0X00 }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X00 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X00 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0X00 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X00 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X00 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X00 }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X00 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X00 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0X00 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X00 }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X00 }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0X00 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0X00 }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0X00 }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X00 }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0X00 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X00 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X00 }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X00 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X00 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0X00 }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0X00 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0X00 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X00 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X00 }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X00 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X00 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X00 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0X00 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X00 }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0X00 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0X00 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0X00 }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0X00 }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X00 }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X00 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X00 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0X00 }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0X00 }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0X00 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0X00 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X00 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0X00 }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X00 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X00 }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0X00 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0X00 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0X00 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X00 }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X00 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X00 }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X00 }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X00 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X00 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X00 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X00 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0X00 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X00 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0X00 }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X00 }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0X00 }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X00 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0X00 }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0X00 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X00 }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0X00 }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X00 }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X00 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X00 }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0X00 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X00 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X00 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0X00 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0X00 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X00 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X00 }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X00 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0X00 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0X00 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X00 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X00 }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X00 }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X00 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X00 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0X00 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X00 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X00 }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0X00 }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X00 }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0X00 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X00 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0X00 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X00 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X00 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X00 }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0X00 }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X00 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X00 }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0X00 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X00 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0X00 }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X00 }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0X00 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X00 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X00 }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0X00 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X00 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X00 }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X00 }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X00 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0X00 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0X00 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0X00 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X00 }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X00 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X00 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X00 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0X00 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X00 }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0X00 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X00 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X00 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X00 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0X00 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X00 }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0X00 }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0X00 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0X00 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X00 }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X00 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0X00 }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X00 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X00 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0X00 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X00 }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0X00 }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X00 }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X00 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X00 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0X00 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X00 }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X00 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0X00 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X00 }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X00 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0X00 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X00 }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X00 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X00 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0X00 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X00 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0X00 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X00 }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X00 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0X00 }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X00 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X00 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X00 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X00 }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0X00 }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0X00 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0X00 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0X00 }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X00 }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X00 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0X00 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0X00 }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0X00 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X00 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X00 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0X00 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0X00 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X00 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X00 }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X00 }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0X00 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X00 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X00 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X00 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0X00 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0X00 }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X00 }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0X00 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X00 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X00 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0X00 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X00 }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X00 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X00 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0X00 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X00 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X00 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X00 }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X00 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0X00 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X00 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X00 }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0X00 }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X00 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X00 }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0X00 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X00 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0X00 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X00 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X00 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X00 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X00 }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X00 }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X00 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X00 }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0X00 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0X00 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X00 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0X00 }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X00 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0X00 }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X00 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X00 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0X00 }, /* 5.5db */ + { 0X7F, 0XB2, 0X61, 0X00 }, /* 6.0db */ +}; +#endif diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c index c078bb0a8437..764304604e2c 100644 --- a/sound/soc/codecs/tas2781-comlib-i2c.c +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -320,6 +320,8 @@ void tasdevice_reset(struct tasdevice_priv *tas_dev) for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, TASDEVICE_REG_SWRESET, + tas_dev->chip_id == TAS5825 ? + TAS5825_REG_SWRESET_RESET : TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index ea3cdb8553de..cf6bd4d63f31 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,8 +30,10 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2x20-tlv.h> #include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include <linux/unaligned.h> #define X2563_CL_STT_VAL(xreg, xval) \ @@ -98,16 +100,26 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { }; static const struct i2c_device_id tasdevice_id[] = { + { "tas2020", TAS2020 }, + { "tas2118", TAS2118 }, + { "tas2120", TAS2120 }, + { "tas2320", TAS2320 }, { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, + { "tas5825", TAS5825 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); #ifdef CONFIG_OF static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2020" }, + { .compatible = "ti,tas2118" }, + { .compatible = "ti,tas2120" }, + { .compatible = "ti,tas2320" }, { .compatible = "ti,tas2563" }, { .compatible = "ti,tas2781" }, + { .compatible = "ti,tas5825" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -797,7 +809,7 @@ static int tasdev_nop_get( return 0; } -static int tas2563_digital_gain_get( +static int tasdevice_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -823,15 +835,15 @@ static int tas2563_digital_gain_get( while (r > 1 + l) { mid = (l + r) / 2; - ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + ar_mid = get_unaligned_be32(tas_dev->dvc_tlv_table[mid]); if (target < ar_mid) r = mid; else l = mid; } - ar_l = get_unaligned_be32(tas2563_dvc_table[l]); - ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + ar_l = get_unaligned_be32(tas_dev->dvc_tlv_table[l]); + ar_r = get_unaligned_be32(tas_dev->dvc_tlv_table[r]); /* find out the member same as or closer to the current volume */ ucontrol->value.integer.value[0] = @@ -841,7 +853,7 @@ static int tas2563_digital_gain_get( return 0; } -static int tas2563_digital_gain_put( +static int tasdevice_digital_gain_put( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -867,7 +879,7 @@ static int tas2563_digital_gain_put( } volrd = get_unaligned_be32(&data[0]); - volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + volwr = get_unaligned_be32(tas_dev->dvc_tlv_table[vol]); if (volrd == volwr) { rc = 0; @@ -876,7 +888,7 @@ static int tas2563_digital_gain_put( for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_bulk_write(tas_dev, i, reg, - (unsigned char *)tas2563_dvc_table[vol], 4); + (unsigned char *)tas_dev->dvc_tlv_table[vol], 4); if (ret) { dev_err(tas_dev->dev, "%s, set digital vol error in dev %d\n", @@ -892,11 +904,6 @@ static int tas2563_digital_gain_put( return rc; } -static const struct snd_kcontrol_new tasdevice_snd_controls[] = { - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tasdev_force_fwload_get, tasdev_force_fwload_put), -}; - static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, tasdev_nop_get, tasdev_calib_stop_put), @@ -907,6 +914,16 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), }; +static const struct snd_kcontrol_new tas2x20_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2X20_AMP_LEVEL, + 0, 0, 42, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas2x20_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2X20_DVC_LEVEL, + 0, 0, ARRAY_SIZE(tas2x20_dvc_table) - 1, 0, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, + tas2x20_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -916,6 +933,15 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, tas2781_dvc_tlv), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_cali_controls[] = { SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), }; @@ -923,7 +949,7 @@ static const struct snd_kcontrol_new tas2781_cali_controls[] = { static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, - tas2563_digital_gain_get, tas2563_digital_gain_put, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, tas2563_dvc_tlv), }; @@ -968,7 +994,7 @@ static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; - uinfo->value.integer.min = TAS2563; + uinfo->value.integer.min = TAS2020; uinfo->value.integer.max = TAS2781; return 0; @@ -1168,9 +1194,9 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *active_dev_num, *chip_id; + char *active_dev_num, *chip_id, *fw_load; char *conf_name, *prog_name; - int nr_controls = 4; + int nr_controls = 5; int mix_index = 0; /* Alloc kcontrol via devm_kzalloc, which don't manually @@ -1228,6 +1254,19 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; + fw_load = devm_kstrdup(tas_priv->dev, "Speaker Force Firmware Load", + GFP_KERNEL); + if (!fw_load) + return -ENOMEM; + + dsp_ctrls[mix_index].name = fw_load; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_info_bool_ext; + dsp_ctrls[mix_index].put = tasdev_force_fwload_put; + dsp_ctrls[mix_index].get = tasdev_force_fwload_get; + dsp_ctrls[mix_index].private_value = 0UL; + mix_index++; + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); } @@ -1587,6 +1626,9 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + /* There is no DSP firmware required for TAS2118/2X20. */ + if (tas_priv->chip_id < TAS2563) + goto out; if (tas_priv->name_prefix) scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); @@ -1608,34 +1650,37 @@ static void tasdevice_fw_ready(const struct firmware *fmw, dev_err(tas_priv->dev, "dsp controls error\n"); goto out; } - - ret = tasdevice_create_cali_ctrls(tas_priv); - if (ret) { - dev_err(tas_priv->dev, "cali controls error\n"); - goto out; - } - tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - /* If calibrated data occurs error, dsp will still works with default - * calibrated data inside algo. - */ - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->name_prefix) - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, - tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - else - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s_cal_0x%02x.bin", tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - ret = tas2781_load_calibration(tas_priv, - tas_priv->cal_binaryname[i], i); - if (ret != 0) - dev_err(tas_priv->dev, - "%s: load %s error, default will effect\n", - __func__, tas_priv->cal_binaryname[i]); + /* There is no calibration required for TAS5825. */ + if (tas_priv->chip_id < TAS5825) { + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + /* If calibrated data occurs error, dsp will still works + * with default calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", + tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, keep default.\n", + __func__, tas_priv->cal_binaryname[i]); + } } tasdevice_prmg_load(tas_priv, 0); @@ -1658,7 +1703,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, comp, &acoustic_ctl_fops); #endif out: - if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + if ((tas_priv->fw_state == TASDEVICE_RCA_FW_OK) && + (tas_priv->chip_id >= TAS2563)) { /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } @@ -1804,13 +1850,27 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) int rc; switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + p = (struct snd_kcontrol_new *)tas2x20_snd_controls; + size = ARRAY_SIZE(tas2x20_snd_controls); + tas_priv->dvc_tlv_table = tas2x20_dvc_table; + break; case TAS2781: p = (struct snd_kcontrol_new *)tas2781_snd_controls; size = ARRAY_SIZE(tas2781_snd_controls); break; + case TAS5825: + p = (struct snd_kcontrol_new *)tas5825_snd_controls; + size = ARRAY_SIZE(tas5825_snd_controls); + break; default: p = (struct snd_kcontrol_new *)tas2563_snd_controls; size = ARRAY_SIZE(tas2563_snd_controls); + tas_priv->dvc_tlv_table = tas2563_dvc_table; + break; } rc = snd_soc_add_component_controls(codec, p, size); @@ -1850,8 +1910,6 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tasdevice_snd_controls, - .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, @@ -1967,7 +2025,13 @@ static void tasdevice_i2c_remove(struct i2c_client *client) #ifdef CONFIG_ACPI static const struct acpi_device_id tasdevice_acpi_match[] = { - { "TAS2781", TAS2781 }, + { "TXNW2020", TAS2020 }, + { "TXNW2118", TAS2118 }, + { "TXNW2120", TAS2120 }, + { "TXNW2320", TAS2320 }, + { "TXNW2563", TAS2563 }, + { "TXNW2781", TAS2781 }, + { "TXNW5825", TAS5825 }, {}, }; -- 2.43.0

1 0

[PATCH] ALSA: hda: intel-dsp-config: Select SOF driver on MTL Chromebooks
by Brady Norander 22 Aug '25

22 Aug '25

The SOF driver is required for functional audio on MTL Chromebooks Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- sound/hda/core/intel-dsp-config.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/hda/core/intel-dsp-config.c b/sound/hda/core/intel-dsp-config.c index 3cb1e7fc3b3b..00f184917623 100644 --- a/sound/hda/core/intel-dsp-config.c +++ b/sound/hda/core/intel-dsp-config.c @@ -167,9 +167,9 @@ static const struct config_entry config_table[] = { /* * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake, AlderLake, - * RaptorLake use legacy HDAudio driver except for Google Chromebooks - * and when DMICs are present. Two cases are required since Coreboot - * does not expose NHLT tables. + * RaptorLake, MeteorLake use legacy HDAudio driver except for Google + * Chromebooks and when DMICs are present. Two cases are required since + * Coreboot does not expose NHLT tables. * * When the Chromebook quirk is not present, it's based on information * that no such device exists. When the quirk is present, it could be @@ -516,6 +516,19 @@ static const struct config_entry config_table[] = { /* Meteor Lake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_METEORLAKE) /* Meteorlake-P */ + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_MTL, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_MTL, -- 2.50.1

1 0

Mute LED not working.
by GitHub issues - opened 21 Aug '25

21 Aug '25

alsa-project/alsa-lib issue #473 was opened from Gropinou: https://alsa-project.org/db/?f=a68feee728b669f547491855254add15fddad73e Issue URL : https://github.com/alsa-project/alsa-lib/issues/473 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.17-rc2
by Mark Brown 21 Aug '25

21 Aug '25

The following changes since commit c17b750b3ad9f45f2b6f7e6f7f4679844244f0b9: Linux 6.17-rc2 (2025-08-17 15:22:10 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-rc2 for you to fetch changes up to 7c15e4cabfa96ed3cd717a2ed7a9961268ab21f7: ASoC: cs35l56: Fixes for CS35L63 for production (2025-08-20 19:56:20 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.17 A few fixes that came in during the past week, there's some updates for the CS35L56 which adjust the driver for production silicon and a fix for buggy resume of the ES9389. ---------------------------------------------------------------- Mark Brown (1): ASoC: cs35l56: Fixes for CS35L63 for production Richard Fitzgerald (1): ASoC: cs35l56: Handle new algorithms IDs for CS35L63 Stefan Binding (2): ASoC: cs35l56: Update Firmware Addresses for CS35L63 for production silicon ASoC: cs35l56: Remove SoundWire Clock Divider workaround for CS35L63 Zhang Yi (1): ASoC: codecs: ES9389: Modify the standby configuration include/sound/cs35l56.h | 5 +-- sound/soc/codecs/cs35l56-sdw.c | 69 --------------------------------------- sound/soc/codecs/cs35l56-shared.c | 29 ++++++++++++++-- sound/soc/codecs/cs35l56.c | 2 +- sound/soc/codecs/cs35l56.h | 3 -- sound/soc/codecs/es8389.c | 2 +- 6 files changed, 31 insertions(+), 79 deletions(-)

1 1

Cybersecurity Risk Assessment Request from Emerson for Advanced Linux Sound Architecture (ALSA)
by KATARE, SAURABH [EMR/MSOL/PUNE] 21 Aug '25

21 Aug '25

Hello, I hope this message finds you well. As part of our ongoing efforts to comply with the EU Cyber Resilience Act (CRA), we are currently conducting a cybersecurity risk assessment of third-party software vendors whose products or components are integrated into our systems. To support this initiative, we kindly request your input on the following questions related to your software product "Advanced Linux Sound Architecture (ALSA)" with version v1.2.1.2. Please provide your responses directly in the table below and do reply to all added in this email, Additional Information: * Purpose: This security assessment is part of our due diligence and regulatory compliance obligations under the EU CRA. * Confidentiality: All information shared will be treated as confidential and used solely for the purpose of this assessment. * Contact: Should you have any questions or need further clarification, please feel free to reach out by replying directly to this email. We kindly request your response by Monday, August 25, 2025, to ensure timely completion of our assessment process. Thank you for your cooperation and continued partnership in maintaining a secure and resilient digital environment. Sr. No. Queries to Vendor Response from Vendor (Yes/No) Additional Remarks from Vendor 1 Is Secure Software Development Lifecycle followed for developing this component? 2 Do you provide regular security updates for "ALSA" ? 3 Is there any discontinuation/End of life for the latest version of "ALSA" in near future? 4 Do you have Long Term support for "ALSA"? If yes please mention the version in Remark column 5 Is appropriate cybersecurity testing followed? If yes, is any specific standard for testing used? 6 Are there any vulnerabilities in the latest version which are not disclosed publicly? If yes, when will it be fixed and released? please mention in Remark column. 7 Is the vulnerability handing procedure available for "ALSA"? if yes mention the procedure in the Remark column. 8 Do you comply with EU-CRA requirements? 9 Do you provide proof of conformity regarding adherence to EU-CRA? If yes please mention details in Remark column Best regards, Saurabh. Saurabh Katare | Engineer, Software Development Emerson | Plot no 23, Rajiv Gandhi InfoTech Park | Phase II , Hinjawadi | Pune | Maharashtra | 411057 | India saurabh.katare(a)emerson.com

2 1

[PATCH v2] ASoC: amd: acp: Adjust pdm gain value
by Venkata Prasad Potturu 21 Aug '25

21 Aug '25

Set pdm gain value by setting PDM_MISC_CTRL_MASK value. To avoid low pdm gain value. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Changes since v1: - remove pdm gain adjustment using module parameter. sound/soc/amd/acp/amd.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index cb8d97122f95..73a028e67246 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -130,7 +130,7 @@ #define PDM_DMA_INTR_MASK 0x10000 #define PDM_DEC_64 0x2 #define PDM_CLK_FREQ_MASK 0x07 -#define PDM_MISC_CTRL_MASK 0x10 +#define PDM_MISC_CTRL_MASK 0x18 #define PDM_ENABLE 0x01 #define PDM_DISABLE 0x00 #define DMA_EN_MASK 0x02 -- 2.43.0

3 2

Low maximum volume with Plantronics Blackwire 315.1
by Paul Menzel 21 Aug '25

21 Aug '25

Dear Linux folks, On the Intel Kaby Lake laptop Dell XPS 13 9360, using Debian sid/unstable with Debian sid/unstable and Linux 6.17.0-rc2-00053-gb19a97d57c15, connecting the USB headset Plantronics Blackwire 315.1, even turning the volume on the headset control (or GNOME, both are linked) to maximum, the volume is still very low. `pavucontrol` shows everything on 100 %. Is that a driver problem, or somehow a hardware property? I uploaded the files done by `/sbin/alsa-info` [1]. Kind regards, Paul [1]: https://alsa-project.org/db/?f=ccc9945a112e88c7b6bf29ecc072f05e0cb72f3f

1 0

[PATCH v1] ALSA: hda/tas2781: Enable init_profile_id for device initialization
by Shenghao Ding 21 Aug '25

21 Aug '25

Optimize the time consumption of profile switching, init_profile saves the common settings of different profiles, such as the dsp coefficients, etc, which can greatly reduce the profile switching time comsumption and remove the repetitive settings. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 318f8c58ae61..97c33fee9660 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -477,6 +477,12 @@ static void tasdevice_dspfw_init(void *context) if (tas_priv->fmw->nr_configurations > 0) tas_priv->cur_conf = 0; + /* Init common setting for different audio profiles */ + if (tas_priv->rcabin.init_profile_id >= 0) + tasdevice_select_cfg_blk(tas_priv, + tas_priv->rcabin.init_profile_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + /* If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. */ @@ -779,6 +785,12 @@ static int tas2781_system_resume(struct device *dev) tasdevice_reset(tas_hda->priv); tasdevice_prmg_load(tas_hda->priv, tas_hda->priv->cur_prog); + /* Init common setting for different audio profiles */ + if (tas_hda->priv->rcabin.init_profile_id >= 0) + tasdevice_select_cfg_blk(tas_hda->priv, + tas_hda->priv->rcabin.init_profile_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + if (tas_hda->priv->playback_started) tasdevice_tuning_switch(tas_hda->priv, 0); -- 2.43.0

3 3

[GIT PULL] ASoC fixes for v6.17-rc2
by Mark Brown 21 Aug '25

21 Aug '25

The following changes since commit c17b750b3ad9f45f2b6f7e6f7f4679844244f0b9: Linux 6.17-rc2 (2025-08-17 15:22:10 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-rc2 for you to fetch changes up to 7c15e4cabfa96ed3cd717a2ed7a9961268ab21f7: ASoC: cs35l56: Fixes for CS35L63 for production (2025-08-20 19:56:20 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.17 A few fixes that came in during the past week, there's some updates for the CS35L56 which adjust the driver for production silicon and a fix for buggy resume of the ES9389. ---------------------------------------------------------------- Mark Brown (1): ASoC: cs35l56: Fixes for CS35L63 for production Richard Fitzgerald (1): ASoC: cs35l56: Handle new algorithms IDs for CS35L63 Stefan Binding (2): ASoC: cs35l56: Update Firmware Addresses for CS35L63 for production silicon ASoC: cs35l56: Remove SoundWire Clock Divider workaround for CS35L63 Zhang Yi (1): ASoC: codecs: ES9389: Modify the standby configuration include/sound/cs35l56.h | 5 +-- sound/soc/codecs/cs35l56-sdw.c | 69 --------------------------------------- sound/soc/codecs/cs35l56-shared.c | 29 ++++++++++++++-- sound/soc/codecs/cs35l56.c | 2 +- sound/soc/codecs/cs35l56.h | 3 -- sound/soc/codecs/es8389.c | 2 +- 6 files changed, 31 insertions(+), 79 deletions(-)

2 1

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - opened 21 Aug '25

21 Aug '25

alsa-project/alsa-lib issue #472 was opened from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from mutliple soundcards on the Raspberry Pi. The results are a garbled mess. I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 20 Aug '25

20 Aug '25

Add tas2020, tas2118, tas2120, tas2320, tas5825 support in tas2781 driver. Tas2118, tas2x20 have no on-chip DSP, tas5825 have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781.h | 11 +- include/sound/tas2x20-tlv.h | 259 ++++++++++++++++++++++++++ sound/soc/codecs/tas2781-comlib-i2c.c | 2 + sound/soc/codecs/tas2781-i2c.c | 156 +++++++++++----- 4 files changed, 381 insertions(+), 47 deletions(-) create mode 100644 include/sound/tas2x20-tlv.h diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index f0aefc04a957..73ac447b9327 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -51,7 +51,9 @@ /* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) + +#define TAS5825_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) @@ -110,8 +112,14 @@ #define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { + TAS2020, + TAS2118, + TAS2120, + TAS2320, TAS2563, TAS2781, + TAS5825, + TAS_OTHERS, }; enum dspbin_type { @@ -194,6 +202,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const unsigned char (*dvc_tlv_table)[4]; const char *name_prefix; unsigned char ndev; unsigned int dspbin_typ; diff --git a/include/sound/tas2x20-tlv.h b/include/sound/tas2x20-tlv.h new file mode 100644 index 000000000000..6e6bcec4a0a1 --- /dev/null +++ b/include/sound/tas2x20-tlv.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2x20/TAS2118 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2x20/TAS2118 hda driver implements for one, two, or even multiple +// TAS2x20/TAS2118 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2X20_TLV_H__ +#define __TAS2X20_TLV_H__ + +#define TAS2X20_DVC_LEVEL TASDEVICE_REG(0x0, 0x2, 0x0c) +#define TAS2X20_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x07) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_amp_tlv, 2100, 50, 0); + +/* pow(10, db/20) * pow(2,22) */ +static const __maybe_unused unsigned char tas2x20_dvc_table[][4] = { + { 0X00, 0X00, 0X0D, 0X00 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0X00 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0X00 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0X00 }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0X00 }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0X00 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X00 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X00 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X00 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0X00 }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X00 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X00 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0X00 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X00 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X00 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X00 }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X00 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X00 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0X00 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X00 }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X00 }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0X00 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0X00 }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0X00 }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X00 }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0X00 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X00 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X00 }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X00 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X00 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0X00 }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0X00 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0X00 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X00 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X00 }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X00 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X00 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X00 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0X00 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X00 }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0X00 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0X00 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0X00 }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0X00 }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X00 }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X00 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X00 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0X00 }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0X00 }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0X00 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0X00 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X00 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0X00 }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X00 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X00 }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0X00 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0X00 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0X00 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X00 }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X00 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X00 }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X00 }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X00 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X00 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X00 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X00 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0X00 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X00 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0X00 }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X00 }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0X00 }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X00 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0X00 }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0X00 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X00 }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0X00 }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X00 }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X00 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X00 }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0X00 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X00 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X00 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0X00 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0X00 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X00 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X00 }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X00 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0X00 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0X00 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X00 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X00 }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X00 }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X00 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X00 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0X00 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X00 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X00 }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0X00 }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X00 }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0X00 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X00 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0X00 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X00 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X00 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X00 }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0X00 }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X00 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X00 }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0X00 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X00 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0X00 }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X00 }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0X00 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X00 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X00 }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0X00 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X00 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X00 }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X00 }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X00 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0X00 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0X00 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0X00 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X00 }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X00 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X00 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X00 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0X00 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X00 }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0X00 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X00 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X00 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X00 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0X00 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X00 }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0X00 }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0X00 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0X00 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X00 }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X00 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0X00 }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X00 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X00 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0X00 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X00 }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0X00 }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X00 }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X00 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X00 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0X00 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X00 }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X00 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0X00 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X00 }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X00 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0X00 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X00 }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X00 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X00 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0X00 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X00 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0X00 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X00 }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X00 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0X00 }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X00 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X00 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X00 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X00 }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0X00 }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0X00 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0X00 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0X00 }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X00 }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X00 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0X00 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0X00 }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0X00 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X00 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X00 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0X00 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0X00 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X00 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X00 }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X00 }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0X00 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X00 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X00 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X00 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0X00 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0X00 }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X00 }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0X00 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X00 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X00 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0X00 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X00 }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X00 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X00 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0X00 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X00 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X00 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X00 }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X00 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0X00 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X00 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X00 }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0X00 }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X00 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X00 }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0X00 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X00 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0X00 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X00 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X00 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X00 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X00 }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X00 }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X00 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X00 }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0X00 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0X00 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X00 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0X00 }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X00 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0X00 }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X00 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X00 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0X00 }, /* 5.5db */ + { 0X7F, 0XB2, 0X61, 0X00 }, /* 6.0db */ +}; +#endif diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c index c078bb0a8437..764304604e2c 100644 --- a/sound/soc/codecs/tas2781-comlib-i2c.c +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -320,6 +320,8 @@ void tasdevice_reset(struct tasdevice_priv *tas_dev) for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, TASDEVICE_REG_SWRESET, + tas_dev->chip_id == TAS5825 ? + TAS5825_REG_SWRESET_RESET : TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index ea3cdb8553de..cf6bd4d63f31 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,8 +30,10 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2x20-tlv.h> #include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include <linux/unaligned.h> #define X2563_CL_STT_VAL(xreg, xval) \ @@ -98,16 +100,26 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { }; static const struct i2c_device_id tasdevice_id[] = { + { "tas2020", TAS2020 }, + { "tas2118", TAS2118 }, + { "tas2120", TAS2120 }, + { "tas2320", TAS2320 }, { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, + { "tas5825", TAS5825 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); #ifdef CONFIG_OF static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2020" }, + { .compatible = "ti,tas2118" }, + { .compatible = "ti,tas2120" }, + { .compatible = "ti,tas2320" }, { .compatible = "ti,tas2563" }, { .compatible = "ti,tas2781" }, + { .compatible = "ti,tas5825" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -797,7 +809,7 @@ static int tasdev_nop_get( return 0; } -static int tas2563_digital_gain_get( +static int tasdevice_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -823,15 +835,15 @@ static int tas2563_digital_gain_get( while (r > 1 + l) { mid = (l + r) / 2; - ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + ar_mid = get_unaligned_be32(tas_dev->dvc_tlv_table[mid]); if (target < ar_mid) r = mid; else l = mid; } - ar_l = get_unaligned_be32(tas2563_dvc_table[l]); - ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + ar_l = get_unaligned_be32(tas_dev->dvc_tlv_table[l]); + ar_r = get_unaligned_be32(tas_dev->dvc_tlv_table[r]); /* find out the member same as or closer to the current volume */ ucontrol->value.integer.value[0] = @@ -841,7 +853,7 @@ static int tas2563_digital_gain_get( return 0; } -static int tas2563_digital_gain_put( +static int tasdevice_digital_gain_put( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -867,7 +879,7 @@ static int tas2563_digital_gain_put( } volrd = get_unaligned_be32(&data[0]); - volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + volwr = get_unaligned_be32(tas_dev->dvc_tlv_table[vol]); if (volrd == volwr) { rc = 0; @@ -876,7 +888,7 @@ static int tas2563_digital_gain_put( for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_bulk_write(tas_dev, i, reg, - (unsigned char *)tas2563_dvc_table[vol], 4); + (unsigned char *)tas_dev->dvc_tlv_table[vol], 4); if (ret) { dev_err(tas_dev->dev, "%s, set digital vol error in dev %d\n", @@ -892,11 +904,6 @@ static int tas2563_digital_gain_put( return rc; } -static const struct snd_kcontrol_new tasdevice_snd_controls[] = { - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tasdev_force_fwload_get, tasdev_force_fwload_put), -}; - static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, tasdev_nop_get, tasdev_calib_stop_put), @@ -907,6 +914,16 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), }; +static const struct snd_kcontrol_new tas2x20_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2X20_AMP_LEVEL, + 0, 0, 42, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas2x20_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2X20_DVC_LEVEL, + 0, 0, ARRAY_SIZE(tas2x20_dvc_table) - 1, 0, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, + tas2x20_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -916,6 +933,15 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, tas2781_dvc_tlv), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_cali_controls[] = { SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), }; @@ -923,7 +949,7 @@ static const struct snd_kcontrol_new tas2781_cali_controls[] = { static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, - tas2563_digital_gain_get, tas2563_digital_gain_put, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, tas2563_dvc_tlv), }; @@ -968,7 +994,7 @@ static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; - uinfo->value.integer.min = TAS2563; + uinfo->value.integer.min = TAS2020; uinfo->value.integer.max = TAS2781; return 0; @@ -1168,9 +1194,9 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *active_dev_num, *chip_id; + char *active_dev_num, *chip_id, *fw_load; char *conf_name, *prog_name; - int nr_controls = 4; + int nr_controls = 5; int mix_index = 0; /* Alloc kcontrol via devm_kzalloc, which don't manually @@ -1228,6 +1254,19 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; + fw_load = devm_kstrdup(tas_priv->dev, "Speaker Force Firmware Load", + GFP_KERNEL); + if (!fw_load) + return -ENOMEM; + + dsp_ctrls[mix_index].name = fw_load; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_info_bool_ext; + dsp_ctrls[mix_index].put = tasdev_force_fwload_put; + dsp_ctrls[mix_index].get = tasdev_force_fwload_get; + dsp_ctrls[mix_index].private_value = 0UL; + mix_index++; + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); } @@ -1587,6 +1626,9 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + /* There is no DSP firmware required for TAS2118/2X20. */ + if (tas_priv->chip_id < TAS2563) + goto out; if (tas_priv->name_prefix) scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); @@ -1608,34 +1650,37 @@ static void tasdevice_fw_ready(const struct firmware *fmw, dev_err(tas_priv->dev, "dsp controls error\n"); goto out; } - - ret = tasdevice_create_cali_ctrls(tas_priv); - if (ret) { - dev_err(tas_priv->dev, "cali controls error\n"); - goto out; - } - tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - /* If calibrated data occurs error, dsp will still works with default - * calibrated data inside algo. - */ - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->name_prefix) - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, - tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - else - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s_cal_0x%02x.bin", tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - ret = tas2781_load_calibration(tas_priv, - tas_priv->cal_binaryname[i], i); - if (ret != 0) - dev_err(tas_priv->dev, - "%s: load %s error, default will effect\n", - __func__, tas_priv->cal_binaryname[i]); + /* There is no calibration required for TAS5825. */ + if (tas_priv->chip_id < TAS5825) { + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + /* If calibrated data occurs error, dsp will still works + * with default calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", + tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, keep default.\n", + __func__, tas_priv->cal_binaryname[i]); + } } tasdevice_prmg_load(tas_priv, 0); @@ -1658,7 +1703,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, comp, &acoustic_ctl_fops); #endif out: - if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + if ((tas_priv->fw_state == TASDEVICE_RCA_FW_OK) && + (tas_priv->chip_id >= TAS2563)) { /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } @@ -1804,13 +1850,27 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) int rc; switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + p = (struct snd_kcontrol_new *)tas2x20_snd_controls; + size = ARRAY_SIZE(tas2x20_snd_controls); + tas_priv->dvc_tlv_table = tas2x20_dvc_table; + break; case TAS2781: p = (struct snd_kcontrol_new *)tas2781_snd_controls; size = ARRAY_SIZE(tas2781_snd_controls); break; + case TAS5825: + p = (struct snd_kcontrol_new *)tas5825_snd_controls; + size = ARRAY_SIZE(tas5825_snd_controls); + break; default: p = (struct snd_kcontrol_new *)tas2563_snd_controls; size = ARRAY_SIZE(tas2563_snd_controls); + tas_priv->dvc_tlv_table = tas2563_dvc_table; + break; } rc = snd_soc_add_component_controls(codec, p, size); @@ -1850,8 +1910,6 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tasdevice_snd_controls, - .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, @@ -1967,7 +2025,13 @@ static void tasdevice_i2c_remove(struct i2c_client *client) #ifdef CONFIG_ACPI static const struct acpi_device_id tasdevice_acpi_match[] = { - { "TAS2781", TAS2781 }, + { "TXNW2020", TAS2020 }, + { "TXNW2118", TAS2118 }, + { "TXNW2120", TAS2120 }, + { "TXNW2320", TAS2320 }, + { "TXNW2563", TAS2563 }, + { "TXNW2781", TAS2781 }, + { "TXNW5825", TAS5825 }, {}, }; -- 2.43.0

1 0

[tiwai-sound:for-next 17/18] sound/usb/usx2y/us144mkii_midi.c:65 tascam_midi_in_urb_complete() error: we previously assumed 'tascam' could be null (see line 54)
by Dan Carpenter 19 Aug '25

19 Aug '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: d9f0633867711cff87c5604a4f90daa8ab986e4f commit: 0ec417aa85467c6fad0563621b74868be0faf00d [17/18] ALSA: usb-audio: Add infrastructure for TASCAM US-144MKII config: powerpc64-randconfig-r073-20250819 (https://download.01.org/0day-ci/archive/20250819/202508192109.lcMrINK1-lkp@…) compiler: clang version 22.0.0git (https://github.com/llvm/llvm-project 93d24b6b7b148c47a2fa228a4ef31524fa1d9f3f) 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> | Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> | Closes: https://lore.kernel.org/r/202508192109.lcMrINK1-lkp@intel.com/ smatch warnings: sound/usb/usx2y/us144mkii_midi.c:65 tascam_midi_in_urb_complete() error: we previously assumed 'tascam' could be null (see line 54) vim +/tascam +65 sound/usb/usx2y/us144mkii_midi.c 67afec157fe63d Šerif Rami 2025-08-14 38 67afec157fe63d Šerif Rami 2025-08-14 39 void tascam_midi_in_urb_complete(struct urb *urb) 67afec157fe63d Šerif Rami 2025-08-14 40 { 67afec157fe63d Šerif Rami 2025-08-14 41 struct tascam_card *tascam = urb->context; 67afec157fe63d Šerif Rami 2025-08-14 42 int ret; 67afec157fe63d Šerif Rami 2025-08-14 43 67afec157fe63d Šerif Rami 2025-08-14 44 if (urb->status) { 67afec157fe63d Šerif Rami 2025-08-14 45 if (urb->status != -ENOENT && urb->status != -ECONNRESET && 67afec157fe63d Šerif Rami 2025-08-14 46 urb->status != -ESHUTDOWN && urb->status != -EPROTO) { 67afec157fe63d Šerif Rami 2025-08-14 47 dev_err_ratelimited(tascam->card->dev, ^^^^^^^^^^^^ Unchecked dereference. 67afec157fe63d Šerif Rami 2025-08-14 48 "MIDI IN URB failed: status %d\n", 67afec157fe63d Šerif Rami 2025-08-14 49 urb->status); 67afec157fe63d Šerif Rami 2025-08-14 50 } 67afec157fe63d Šerif Rami 2025-08-14 51 goto out; 67afec157fe63d Šerif Rami 2025-08-14 52 } 67afec157fe63d Šerif Rami 2025-08-14 53 67afec157fe63d Šerif Rami 2025-08-14 @54 if (tascam && atomic_read(&tascam->midi_in_active) && ^^^^^^ Is this required? 67afec157fe63d Šerif Rami 2025-08-14 55 urb->actual_length > 0) { 67afec157fe63d Šerif Rami 2025-08-14 56 kfifo_in_spinlocked(&tascam->midi_in_fifo, urb->transfer_buffer, 67afec157fe63d Šerif Rami 2025-08-14 57 urb->actual_length, &tascam->midi_in_lock); 67afec157fe63d Šerif Rami 2025-08-14 58 schedule_work(&tascam->midi_in_work); 67afec157fe63d Šerif Rami 2025-08-14 59 } 67afec157fe63d Šerif Rami 2025-08-14 60 67afec157fe63d Šerif Rami 2025-08-14 61 usb_get_urb(urb); 67afec157fe63d Šerif Rami 2025-08-14 62 usb_anchor_urb(urb, &tascam->midi_in_anchor); ^^^^^^^^ Unchecked dereference. 67afec157fe63d Šerif Rami 2025-08-14 63 ret = usb_submit_urb(urb, GFP_ATOMIC); 67afec157fe63d Šerif Rami 2025-08-14 64 if (ret < 0) { 67afec157fe63d Šerif Rami 2025-08-14 @65 dev_err(tascam->card->dev, 67afec157fe63d Šerif Rami 2025-08-14 66 "Failed to resubmit MIDI IN URB: error %d\n", ret); 67afec157fe63d Šerif Rami 2025-08-14 67 usb_unanchor_urb(urb); 67afec157fe63d Šerif Rami 2025-08-14 68 usb_put_urb(urb); 67afec157fe63d Šerif Rami 2025-08-14 69 } 67afec157fe63d Šerif Rami 2025-08-14 70 out: 67afec157fe63d Šerif Rami 2025-08-14 71 usb_put_urb(urb); 67afec157fe63d Šerif Rami 2025-08-14 72 } -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

Bug 220109 - Audio disappears on HP 15-fc000 after warm boot again
by Kailang 19 Aug '25

19 Aug '25

Hi Takashi, Attach patch will solve issue. BR, Kailang

2 1

[PATCH v2] ALSA: hda/tas2781: Add name prefix tas2781 for tas2781's dvc_tlv and amp_vol_tlv
by Shenghao Ding 18 Aug '25

18 Aug '25

With some new devices adding into the driver, dvc_tlv and amp_vol_tlv will cause confusion for customers on which devices they support. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Put Fixes tag. --- include/sound/tas2781-tlv.h | 6 +++--- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- sound/hda/codecs/side-codecs/tas2781_hda_spi.c | 6 ++++-- sound/soc/codecs/tas2781-i2c.c | 4 ++-- 4 files changed, 10 insertions(+), 8 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index ef9b9f19d212..273224df9282 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 50, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_dvc_tlv, -10000, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 3471b9ab52cb..318f8c58ae61 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -269,7 +269,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c index 09a5d0f131b2..b9a55672bf15 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c @@ -494,9 +494,11 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new tas2781_snd_ctls[] = { ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_AMP_LEVEL, 1, 0, 20, 0, - tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_getvol, tas2781_amp_putvol, + tas2781_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_DVC_LVL, 0, 0, 200, 1, - tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), + tas2781_digital_getvol, tas2781_digital_putvol, + tas2781_dvc_tlv), ACARD_SINGLE_BOOL_EXT(NULL, 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e4ca43e006db..ea3cdb8553de 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -910,10 +910,10 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, - tas2781_digital_putvol, dvc_tlv), + tas2781_digital_putvol, tas2781_dvc_tlv), }; static const struct snd_kcontrol_new tas2781_cali_controls[] = { -- 2.43.0

2 1

[PATCH v1] ALSA: hda/tas2781: Add name prefix tas2781 for tas2781's dvc_tlv and amp_vol_tlv
by Shenghao Ding 15 Aug '25

15 Aug '25

With some new devices adding into the driver, dvc_tlv and amp_vol_tlv will cause confusion for customers on which devices they support. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781-tlv.h | 6 +++--- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- sound/hda/codecs/side-codecs/tas2781_hda_spi.c | 6 ++++-- sound/soc/codecs/tas2781-i2c.c | 4 ++-- 4 files changed, 10 insertions(+), 8 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index ef9b9f19d212..273224df9282 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 50, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_dvc_tlv, -10000, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 3471b9ab52cb..318f8c58ae61 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -269,7 +269,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c index 09a5d0f131b2..b9a55672bf15 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c @@ -494,9 +494,11 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new tas2781_snd_ctls[] = { ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_AMP_LEVEL, 1, 0, 20, 0, - tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_getvol, tas2781_amp_putvol, + tas2781_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_DVC_LVL, 0, 0, 200, 1, - tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), + tas2781_digital_getvol, tas2781_digital_putvol, + tas2781_dvc_tlv), ACARD_SINGLE_BOOL_EXT(NULL, 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e4ca43e006db..ea3cdb8553de 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -910,10 +910,10 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, - tas2781_digital_putvol, dvc_tlv), + tas2781_digital_putvol, tas2781_dvc_tlv), }; static const struct snd_kcontrol_new tas2781_cali_controls[] = { -- 2.43.0

2 1

[PATCH v1] ALSA: hda/tas2781: Normalize the volume kcontrol name
by Baojun Xu 14 Aug '25

14 Aug '25

Change the name of the kcontrol from "Gain" to "Volume". Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 92aae19cfc8f..e4bc3bc756b0 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -256,7 +256,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { }; static const struct snd_kcontrol_new tas2781_snd_controls[] = { - ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, -- 2.43.0

3 4

[GIT PULL] ASoC fixes for v6.17-rc1
by Mark Brown 14 Aug '25

14 Aug '25

The following changes since commit 8f5ae30d69d7543eee0d70083daf4de8fe15d585: Linux 6.17-rc1 (2025-08-10 19:41:16 +0300) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-rc1 for you to fetch changes up to 47ed64db8c17eb16541098add865178fb7e68744: ASoC: tas2781: Normalize the volume kcontrol name (2025-08-13 15:52:53 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.17 A reasonably small collection of fixes that came in since the merge window, mostly small and driver specific plus a cleanup of the menu reorganisation to address some user confusion with the way the generic drivers had been handled. ---------------------------------------------------------------- Alexey Klimov (1): ASoC: codecs: tx-macro: correct tx_macro_component_drv name Baojun Xu (1): ASoC: tas2781: Normalize the volume kcontrol name Colin Ian King (2): ASoC: codec: sma1307: replace spelling mistake with new error message ASoC: tas2781: Fix spelling mistake "dismatch" -> "mismatch" Kuninori Morimoto (1): ASoC: generic: tidyup standardized ASoC menu for generic Sergio Perez Gonzalez (1): ASoC: stm: stm32_i2s: Fix calc_clk_div() error handling in determine_rate() Shengjiu Wang (1): ASoC: fsl_sai: replace regmap_write with regmap_update_bits Shuming Fan (2): ASoC: rt721: fix FU33 Boost Volume control not working ASoC: rt1320: fix random cycle mute issue Thorsten Blum (1): ASoC: codecs: Call strscpy() with correct size argument sound/soc/Kconfig | 4 +++- sound/soc/codecs/aw87390.c | 8 +++----- sound/soc/codecs/aw88081.c | 5 ++--- sound/soc/codecs/aw88166.c | 8 +++----- sound/soc/codecs/aw88261.c | 8 +++----- sound/soc/codecs/aw88395/aw88395.c | 8 +++----- sound/soc/codecs/aw88399.c | 8 +++----- sound/soc/codecs/lpass-tx-macro.c | 2 +- sound/soc/codecs/rt1320-sdw.c | 3 ++- sound/soc/codecs/rt721-sdca.c | 2 ++ sound/soc/codecs/rt721-sdca.h | 4 ++++ sound/soc/codecs/sma1307.c | 2 +- sound/soc/codecs/tas2781-i2c.c | 6 +++--- sound/soc/fsl/fsl_sai.c | 20 ++++++++++---------- sound/soc/stm/stm32_i2s.c | 7 ++----- 15 files changed, 45 insertions(+), 50 deletions(-)

2 1

[PATCH v1] ASoC: tas2781: Normalize the volume kcontrol name
by Baojun Xu 13 Aug '25

13 Aug '25

Change the name of the kcontrol from "Gain" to "Volume". Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index aa5f56615dad..e4ca43e006db 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -908,10 +908,10 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { }; static const struct snd_kcontrol_new tas2781_snd_controls[] = { - SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), - SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), }; -- 2.43.0

1 1

[PATCH v1] ASoC: tas2781: Normalize the volume kcontrol name
by Baojun Xu 13 Aug '25

13 Aug '25

As more devices become supported, add "tas2781" as a suffix before the TLV name, correct the DVC step, and normalize the volume kcontrol name. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781-tlv.h | 6 ++++-- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 4 ++-- sound/hda/codecs/side-codecs/tas2781_hda_spi.c | 5 +++-- sound/soc/codecs/tas2781-i2c.c | 8 ++++---- 4 files changed, 13 insertions(+), 10 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index d87263e43fdb..38b8f828a353 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -15,7 +15,9 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_dvc_tlv, + -10000, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_amp_tlv, + 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index a0b132681804..6ec93b27aa7f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -265,9 +265,9 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { }; static const struct snd_kcontrol_new tas2781_snd_controls[] = { - ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c index 09a5d0f131b2..b4664a696bad 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c @@ -494,9 +494,10 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new tas2781_snd_ctls[] = { ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_AMP_LEVEL, 1, 0, 20, 0, - tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_getvol, tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_DVC_LVL, 0, 0, 200, 1, - tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), + tas2781_digital_getvol, tas2781_digital_putvol, + tas2781_dvc_tlv), ACARD_SINGLE_BOOL_EXT(NULL, 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9f4d965a1335..a51f20cd869a 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -908,12 +908,12 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { }; static const struct snd_kcontrol_new tas2781_snd_controls[] = { - SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), - SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, + tas2781_amp_putvol, tas2781_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, - tas2781_digital_putvol, dvc_tlv), + tas2781_digital_putvol, tas2781_dvc_tlv), }; static const struct snd_kcontrol_new tas2781_cali_controls[] = { -- 2.43.0

2 3

ucm2: Qualcomm: x1e80100: Add X1E001DE-DEVKIT configuration
by GitHub pull_request - opened 13 Aug '25

13 Aug '25

alsa-project/alsa-ucm-conf pull request #602 was opened from stephan-gh: The X1E001DE Devkit needs basically the same audio configuration as the X1E80100 CRD, so just include the HiFi.conf intended for the CRD. The only difference is that there are no speakers connected to the pin header on the mainboard by default, so remove the speakers from the available outputs. If someone wants to connect speakers there they could drop the RemoveDevice line after verifying that it works correctly for their speakers. At the moment we do not have any way to automatically detect if there are speakers connected or not. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/602 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/602.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND PATCH v1] ASoC: tas2781: Add keyword "init" in profile section
by Shenghao Ding 12 Aug '25

12 Aug '25

Since version 0x105, the keyword 'init' was introduced into the profile, which is used for chip initialization, particularly to store common settings for other non-initialization profiles. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781-dsp.h | 8 ++++++++ sound/soc/codecs/tas2781-fmwlib.c | 12 ++++++++++++ sound/soc/codecs/tas2781-i2c.c | 6 ++++++ 3 files changed, 26 insertions(+) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index c3a9efa73d5d..a21f34c0266e 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -198,6 +198,14 @@ struct tasdevice_rca { int ncfgs; struct tasdevice_config_info **cfg_info; int profile_cfg_id; + /* + * Since version 0x105, the keyword 'init' was introduced into the + * profile, which is used for chip initialization, particularly to + * store common settings for other non-initialization profiles. + * if (init_profile_id < 0) + * No init profile inside the RCA firmware. + */ + int init_profile_id; }; void tasdevice_select_cfg_blk(void *context, int conf_no, diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index c9c1e608ddb7..8baf56237624 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -180,6 +180,16 @@ static struct tasdevice_config_info *tasdevice_add_config( dev_err(tas_priv->dev, "add conf: Out of boundary\n"); goto out; } + /* If in the RCA bin file are several profiles with the + * keyword "init", init_profile_id only store the last + * init profile id. + */ + if (strnstr(&config_data[config_offset], "init", 64)) { + tas_priv->rcabin.init_profile_id = + tas_priv->rcabin.ncfgs - 1; + dev_dbg(tas_priv->dev, "%s: init profile id = %d\n", + __func__, tas_priv->rcabin.init_profile_id); + } config_offset += 64; } @@ -283,6 +293,8 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) int i; rca = &(tas_priv->rcabin); + /* Initialize to none */ + rca->init_profile_id = -1; fw_hdr = &(rca->fw_hdr); if (!fmw || !fmw->data) { dev_err(tas_priv->dev, "Failed to read %s\n", diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9f4d965a1335..f8faeffd983d 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1641,6 +1641,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; + /* Init common setting for different audio profiles */ + if (tas_priv->rcabin.init_profile_id >= 0) + tasdevice_select_cfg_blk(tas_priv, + tas_priv->rcabin.init_profile_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + #ifdef CONFIG_SND_SOC_TAS2781_ACOUST_I2C if (tas_priv->name_prefix) acoustic_debugfs_node = devm_kasprintf(tas_priv->dev, -- 2.43.0

2 1

[PATCH v1] ASoC: tas2781: Add keyword "init" in profile section
by Shenghao Ding 12 Aug '25

12 Aug '25

Since version 0x105, the keyword 'init' was introduced into the profile, which is used for chip initialization, particularly to store common settings for other non-initialization profiles. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781-dsp.h | 8 ++++++++ sound/soc/codecs/tas2781-fmwlib.c | 12 ++++++++++++ sound/soc/codecs/tas2781-i2c.c | 6 ++++++ 3 files changed, 26 insertions(+) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index c3a9efa73d5d..a21f34c0266e 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -198,6 +198,14 @@ struct tasdevice_rca { int ncfgs; struct tasdevice_config_info **cfg_info; int profile_cfg_id; + /* + * Since version 0x105, the keyword 'init' was introduced into the + * profile, which is used for chip initialization, particularly to + * store common settings for other non-initialization profiles. + * if (init_profile_id < 0) + * No init profile inside the RCA firmware. + */ + int init_profile_id; }; void tasdevice_select_cfg_blk(void *context, int conf_no, diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index c9c1e608ddb7..8baf56237624 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -180,6 +180,16 @@ static struct tasdevice_config_info *tasdevice_add_config( dev_err(tas_priv->dev, "add conf: Out of boundary\n"); goto out; } + /* If in the RCA bin file are several profiles with the + * keyword "init", init_profile_id only store the last + * init profile id. + */ + if (strnstr(&config_data[config_offset], "init", 64)) { + tas_priv->rcabin.init_profile_id = + tas_priv->rcabin.ncfgs - 1; + dev_dbg(tas_priv->dev, "%s: init profile id = %d\n", + __func__, tas_priv->rcabin.init_profile_id); + } config_offset += 64; } @@ -283,6 +293,8 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) int i; rca = &(tas_priv->rcabin); + /* Initialize to none */ + rca->init_profile_id = -1; fw_hdr = &(rca->fw_hdr); if (!fmw || !fmw->data) { dev_err(tas_priv->dev, "Failed to read %s\n", diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9f4d965a1335..f8faeffd983d 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1641,6 +1641,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; + /* Init common setting for different audio profiles */ + if (tas_priv->rcabin.init_profile_id >= 0) + tasdevice_select_cfg_blk(tas_priv, + tas_priv->rcabin.init_profile_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + #ifdef CONFIG_SND_SOC_TAS2781_ACOUST_I2C if (tas_priv->name_prefix) acoustic_debugfs_node = devm_kasprintf(tas_priv->dev, -- 2.43.0

2 1

[PATCH v1] ALSA: hda: Add TAS5825 support
by Baojun Xu 11 Aug '25

11 Aug '25

Add TAS5825 support in TI's HDA driver. TAS5825 is an on-chip DSP, but no calibration is required, and no global address support smart amplifier devices. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781-dsp.h | 3 + include/sound/tas2781.h | 4 +- include/sound/tas5825-tlv.h | 24 +++++++ .../hda/codecs/side-codecs/tas2781_hda_i2c.c | 28 ++++++++ sound/soc/codecs/tas2781-fmwlib.c | 67 ++++++++++++++++++- 5 files changed, 121 insertions(+), 5 deletions(-) create mode 100644 include/sound/tas5825-tlv.h diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index c3a9efa73d5d..49bbf24d6559 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -34,6 +34,7 @@ #define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 #define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 #define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 +#define PPC3_VERSION_TAS5825_BASE 0x114200 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -53,6 +54,8 @@ enum tasdevice_dsp_dev_idx { TASDEVICE_DSP_TAS_2781_DUAL_MONO, TASDEVICE_DSP_TAS_2781_21, TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_5825_MONO, + TASDEVICE_DSP_TAS_5825_DUAL, TASDEVICE_DSP_TAS_MAX_DEVICE }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 3875e92f1ec5..f0aefc04a957 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -49,9 +49,9 @@ #define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ (page * 128)) + reg) -/* Software Reset */ +/* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET BIT(0) +#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) diff --git a/include/sound/tas5825-tlv.h b/include/sound/tas5825-tlv.h new file mode 100644 index 000000000000..95f2d3fad120 --- /dev/null +++ b/include/sound/tas5825-tlv.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS5825 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS5825 hda driver implements for one or two TAS5825 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS5825_TLV_H__ +#define __TAS5825_TLV_H__ + +#define TAS5825_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x4c) +#define TAS5825_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x54) + +static const __maybe_unused DECLARE_TLV_DB_SCALE( + tas5825_dvc_tlv, -10300, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE( + tas5825_amp_tlv, -1550, 50, 0); + +#endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 45ac5e41bd4f..71ee946eb748 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -26,6 +26,7 @@ #include <sound/tlv.h> #include <sound/tas2770-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include "hda_local.h" #include "hda_auto_parser.h" @@ -50,6 +51,7 @@ enum device_chip_id { HDA_TAS2563, HDA_TAS2770, HDA_TAS2781, + HDA_TAS5825, HDA_OTHERS }; @@ -272,6 +274,17 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_force_fwload_get, tas2781_force_fwload_put), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + static const struct snd_kcontrol_new tasdevice_prof_ctrl = { .name = "Speaker Profile Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, @@ -501,6 +514,12 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) ARRAY_SIZE(tas2781_snd_controls)); tasdevice_dspfw_init(context); break; + case HDA_TAS5825: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas5825_snd_controls[0], + ARRAY_SIZE(tas5825_snd_controls)); + tasdevice_dspfw_init(context); + break; case HDA_TAS2563: tasdevice_dspfw_init(context); break; @@ -628,6 +647,7 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2781:00-tas2781-hda.0")) { device_name = "TXNW2781"; + hda_priv->hda_chip_id = HDA_TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else if (strstr(dev_name(&clt->dev), "INT8866")) { @@ -639,6 +659,13 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) hda_priv->hda_chip_id = HDA_TAS2563; hda_priv->save_calibration = tas2563_save_calibration; tas_hda->priv->global_addr = TAS2563_GLOBAL_ADDR; + } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW5825")) { + /* + * TAS5825, integrated on-chip DSP without + * global I2C address and calibration supported. + */ + device_name = "TXNW5825"; + hda_priv->hda_chip_id = HDA_TAS5825; } else { return -ENODEV; } @@ -775,6 +802,7 @@ static const struct acpi_device_id tas2781_acpi_hda_match[] = { {"TIAS2781", 0 }, {"TXNW2770", 0 }, {"TXNW2781", 0 }, + {"TXNW5825", 0 }, {} }; MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index c9c1e608ddb7..d69faef8a4d7 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -91,7 +91,7 @@ struct blktyp_devidx_map { }; static const char deviceNumber[TASDEVICE_DSP_TAS_MAX_DEVICE] = { - 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4 + 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4, 1, 2 }; /* fixed m68k compiling issue: mapping table can save code field */ @@ -509,6 +509,56 @@ static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, return offset; } +static int fw_parse_tas5825_program_data_kernel( + struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &(tas_fmw->programs[i]); + if (offset + 72 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + /* Skip 65 unused byts*/ + offset += 65; + offset = fw_parse_data_kernel(tas_fmw, &(program->dev_data), + fmw, offset); + if (offset < 0) + return offset; + } + + return offset; +} + +static int fw_parse_tas5825_configuration_data_kernel( + struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + struct tasdevice_config *config; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (offset + 80 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + /* Skip extra 8 bytes*/ + offset += 72; + offset = fw_parse_data_kernel(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + return offset; + } + + return offset; +} + static int fw_parse_program_data_kernel( struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) @@ -1826,7 +1876,8 @@ static void dspbin_type_check(struct tasdevice_priv *tas_priv, else tas_priv->dspbin_typ = TASDEV_ALPHA; } - if (tas_priv->dspbin_typ != TASDEV_BASIC) + if ((tas_priv->dspbin_typ != TASDEV_BASIC) && + (ppcver < PPC3_VERSION_TAS5825_BASE)) tas_priv->fw_parse_fct_param_address = fw_parse_fct_param_address; } @@ -1837,7 +1888,17 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION_BASE) { + if (ppcver >= PPC3_VERSION_TAS5825_BASE) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_tas5825_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_tas5825_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + dspbin_type_check(tas_priv, ppcver); + } else if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = -- 2.43.0

1 0

ucm2: Qualcomm: add Radxa Dragon Q6A
by GitHub pull_request - opened 10 Aug '25

10 Aug '25

alsa-project/alsa-ucm-conf pull request #601 was opened from strongtz: This is an upcoming SBC (single board computer) based on QCS6490. Full specs: - Headphone jack using WCD9380 codec - Onboard DisplayPort to HDMI bridge using Radxa RA620 - MI2S0 available via the 40-Pin GPIO header (unused by default) - AMICs using WCD9380 codec via a dedicated MIC header (unused by default) The kernel device tree part is not yet but going to be upstreamed soon. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/601 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/601.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[HDA][HP EliteDesk 800 G5][8086:a348] – HDMI audio missing after BIOS update
by Hermanns Cyrille 09 Aug '25

09 Aug '25

Hello ALSA team, On an HP EliteDesk 800 G5 Desktop Mini with BIOS R21 Ver. 02.22.00, audio controller Intel Cannon Lake PCH cAVS [8086:a348], HDMI audio outputs are no longer detected after this BIOS update. System: - Linux Mint 22.1 Cinnamon 6.4.8 - Kernel: 6.8.0-71-generic - Codecs: Conexant CX20632 (analog), Intel Kabylake HDMI (digital) Issue: After boot, HDMI outputs appear as “Not connected”. HDMI audio works only if I use hdajackretask (Intel Kabylake HDMI → Show unconnected pins → set to HDMI/DisplayPort) and select “Install boot override”. ALSA report: http://alsa-project.org/db/?f=a1ee220062b23ad5c13c6d38cde3f1d046ce6260 Request: Would it be possible to add a quirk for this model so HDMI pins are enabled automatically at boot? Thank you for your help.

1 0

Problems with "snd_pcm_hw_params_set_rate_near" and "snd_pcm_hw_params_set_period_size_near" in x86_64 assembly.
by GitHub issues - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-lib issue #471 was opened from sc1ntilla: ### **PREAMBLE:** I have never worked with any sound system before, so I'm almost sure that it's just me doing something wrong, but I can't see what, so I'm asking for help. I'm doing this project for fun, and to learn a lot of stuff, so feel free to give your opinions / advice if you want to. ### **WHAT HAPPENS:** When calling `snd_pcm_hw_params_set_rate_near` and `snd_pcm_hw_params_set_period_size_near` memory protection violation is issued. I find that strange, because `snd_pcm_hw_params_set_buffer_size_near` works perfectly fine, and it's similiar to these two. ### **WHAT I TRIED:** - Making sure that stack after call is on address divisible by 96 (by adding 8 to stack pointer), - Making sure that `*val` argument is on address divisible by 96, - Misaligning stack by 8, 16, 24 and 32 before calling these functions (by subtracting from stack pointer), - Providing 0 to `*val` argument, to rule out that writing to output crashes, - Providing valid output address to `*dir` argument to rule out that 0 is causing crashes, - Calling these functions in different order, - Trying to call `snd_pcm_set_params` instead, but it says _"Channels count (0) not available for PLAYBACK: Invalid argument"_ however, I'm sure that I provided a valid number (2). I even hardcoded it. But to no avail. Is it my misunderstanding of how alsa-lib works? Very likely. Is my assembly code faulty? Quite possible. I just can't see where the problem lies. If you see error somewhere, please let me know. ### **ADDITIONAL INFORMATION:** I have multiple audio outputs: monitors with built-in speakers, standalone speakers, and analog jack headphones. I hardcoded `hw:2,0` as device name to open. Sometimes it results in `snd_pcm_open` returning _"File descriptor in bad state"_. When I tried using `sysdefault` it often returns _"ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.14/work/alsa-lib-1.2.14/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave"_ Below is .data section in which globals are stored, and source of function that will be used to play audio buffers. .data section: ``` .section .data .align 32 test2: .8byte 0 test: .8byte 0 SND_PCM_NONBLOCK: .8byte 0x1 SND_PCM_ASYNC: .8byte 0x2 SND_PCM_STREAM_PLAYBACK: .8byte 0 SND_PCM_FORMAT_U8: .8byte 1 SND_PCM_FORMAT_U16_LE: .8byte 4 SND_PCM_FORMAT_U24_LE: .8byte 8 SND_PCM_FORMAT_U32_LE: .8byte 12 SND_PCM_FORMAT_S8: .8byte 0 SND_PCM_FORMAT_S16_LE: .8byte 2 SND_PCM_FORMAT_S24_LE: .8byte 6 SND_PCM_FORMAT_S32_LE: .8byte 10 SND_PCM_ACCESS_MMAP_INTERLEAVED: .8byte 0 SND_PCM_STATE_PREPARED: .8byte 2 AUDIO_BUFFER_STATE_IDLE: .8byte 0 AUDIO_BUFFER_STATE_WRITING: .8byte 1 AUDIO_BUFFER_STATE_PLAYING: .8byte 2 t1: .8byte 0 t2: .8byte 0 alsa: alsa.pcm_handle: .8byte 0 alsa.hw_params: .8byte 0 alsa.last_error: .8byte 0 alsa.initialized: .8byte 0 alsa.device_name: .ascii "hw:2,0\0" audio_buffers.count_max: .8byte 16 amd64_audio_last_error: .8byte 0 ``` function: ``` amd64_audio_play: /* rdi: (input) address to audio data: * (+0/2) audio format (1: PCM, 2: IEEE754) * (+2/2) amount of channels * (+4/4) rate in hertz * (+8/4) bytes per sec (frequency * bytes per block) * (+12/2) bytes per block (number of channels * bits per sample / 8) * (+14/2) bits per sample * (+16/8) total size of samples data * (+24/8) address to samples data */ /* Align stack on 96byte boundary: */ push %r15 mov %rsp, %r15 and $-32, %rsp mov %rsp, %rax mov $96, %r14 xor %rdx, %rdx div %r14 sub %rdx, %rsp sub $96, %rsp mov %r15, (%rsp) /* Store registers: */ sub $192, %rsp call amd64_util_store_gpr_regs /* Create stack space: */ sub $192, %rsp mov %rdi, (%rsp) /* stack layout: * (+0/8) rdi (address to audio data) * (+8/8) buffer size (altered by *buffer_size_near) * (+16/8) period size (set when setting period size) * (+24/8) rate in hertz * (+32/8) temp */ xor %r15, %r15 mov 16(%rdi), %r15 mov %r15, 8(%rsp) xor %r15, %r15 mov 4(%rdi), %r15d mov %r15d, 24(%rsp) movq $0, 32(%rsp) /* Open pcm if needed: */ cmpq $0, (alsa.pcm_handle) jne .amd64_audio_play.pcm_is_open lea alsa.pcm_handle, %rdi mov %rdi, (%rdi) lea alsa.device_name, %rsi mov (SND_PCM_ACCESS_MMAP_INTERLEAVED), %rdx mov (SND_PCM_NONBLOCK), %r10 call snd_pcm_open cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_open .amd64_audio_play.pcm_is_open: /* Allocate hw params: */ cmpq $0, (alsa.hw_params) jne .amd64_audio_play.hw_params_are_allocated lea alsa.hw_params, %rdi mov %rdi, (%rdi) call snd_pcm_hw_params_malloc cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_malloc .amd64_audio_play.hw_params_are_allocated: /* Get all possible combinations into hw params: */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi call snd_pcm_hw_params_any cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_any /* Set access: */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov (SND_PCM_ACCESS_MMAP_INTERLEAVED), %rdx call snd_pcm_hw_params_set_access cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_access /* Set format: */ xor %rdi, %rdi mov (%rsp), %rdi mov 14(%rdi), %r15w mov (%rdi), %r14w cmp $1, %r14w jne .amd64_audio_play.error.format_unsupported cmpw $32, %r15w je .amd64_audio_play.format32 cmpw $24, %r15w je .amd64_audio_play.format24 cmpw $16, %r15w je .amd64_audio_play.format16 cmpw $8, %r15w je .amd64_audio_play.format8 jmp .amd64_audio_play.error.format_unsupported .amd64_audio_play.format32: mov (SND_PCM_FORMAT_S32_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format24: mov (SND_PCM_FORMAT_S24_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format16: mov (SND_PCM_FORMAT_S16_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format8: mov (SND_PCM_FORMAT_S8), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format_end: mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi call snd_pcm_hw_params_set_format cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_format /* Set channels: */ xor %rdx, %rdx mov (%rsp), %r15 mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov 2(%r15), %dx call snd_pcm_hw_params_set_channels cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_channels /* Set buffer size near: */ /* Returns 0, meaning it succeeded. No crashes generated. */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $8, %rdx mov $0, %r10 call snd_pcm_hw_params_set_buffer_size_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_buffer_size_near /* Set rate near: */ /* CRASHES */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $24, %rdx mov $0, %r10 call snd_pcm_hw_params_set_rate_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_rate_near /* Set period size near: */ /* CRASHES */ mov 8(%rsp), %rax mov $4, %r15 xor %rdx, %rdx div %r15 mov %rax, 16(%rsp) mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $16, %rdx mov $0, %r10 call snd_pcm_hw_params_set_period_size_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_period_size_near /* Free hw params: */ cmpq $0, (alsa.hw_params) je .amd64_audio_play.hw_params_not_allocated mov (alsa.hw_params), %rdi call snd_pcm_hw_params_free movq $0, (alsa.hw_params) .amd64_audio_play.hw_params_not_allocated: /* Register errors: */ xor %rax, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_open: mov %rax, (amd64_audio_last_error) mov $-1, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_malloc: mov %rax, (amd64_audio_last_error) mov $-2, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_any: mov %rax, (amd64_audio_last_error) mov $-3, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_access: mov %rax, (amd64_audio_last_error) mov $-4, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.format_unsupported: mov $-5, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_format: mov %rax, (amd64_audio_last_error) mov $-6, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_channels: mov %rax, (amd64_audio_last_error) mov $-7, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_buffer_size_near: mov %rax, (amd64_audio_last_error) mov $-8, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_period_size_near: mov %rax, (amd64_audio_last_error) mov $-9, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_rate_near: mov %rax, (amd64_audio_last_error) mov $-10, %rax jmp .amd64_audio_play.exit /* Exit: */ .amd64_audio_play.exit: /* Print error to stdout: */ mov %rax, %rdi call amd64_util_print_dec call amd64_util_print_newline cmpq $0, (amd64_audio_last_error) je .amd64_audio_play.no_error mov (amd64_audio_last_error), %rdi call snd_strerror mov %rax, %rdi call amd64_util_print2 call amd64_util_print_newline .amd64_audio_play.no_error: /* Restore stack: */ add $192, %rsp /* Restore registers without rax: */ mov %rax, 128(%rsp) call amd64_util_restore_gpr_regs mov 128(%rsp), %rax add $192, %rsp /* Restore stack alignment: */ pop %rsp pop %r15 ret ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/471 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Update hdajackretask.desktop
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was edited from trebmuh: - adds a FR comment - change the "GenericName" to "Comment" - adds a Icon= field (closes #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Update hdajackretask.desktop
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was edited from trebmuh: - adds a FR comment - change the "genericname" to "Comment" - adds a Icon= field (closes #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Update hdajackretask.desktop
by GitHub pull_request - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was opened from trebmuh: - adds a FR comment - change the "genericname" to "Comment" - adds a Icon= field (close #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

ucm2: Qualcomm: Add MONACO-EVK HiFi config
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was edited from mohsRafi: Add UCM2 configs for the Qualcomm MONACO-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Qualcomm: Add MONACO-EVK HiFi config
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was edited from mohsRafi: Add UCM2 configs for the Qualcomm MONACO-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Qualcomm: Add MONACO-EVK HiFi config
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was edited from mohsRafi: Add UCM2 configs for the Qualcomm QCS9075-IQ8-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Rename qcs6490-rb3gen2 and qcs9075-iq-evk ucm2 conf
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #596 was edited from mohsRafi: Rename the ucm2 conf for Qualcomm qcs6490-rb3gen2 and qcs9075-iq-evk qcs6490-rb3gen2-snd-card.conf -> QCS6490-RB3Gen2.conf qcs9075-iq-evk-snd-card.conf -> LEMANS-EVK.conf Removed snd-card tags from conf files Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/596 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/596.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire: add rt721 ucm support
by GitHub pull_request - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #600 was opened from jack-cy-yu: Add support for rt721's UCM. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/600 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/600.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.17-merge-window
by Mark Brown 06 Aug '25

06 Aug '25

The following changes since commit e95122a32e777309412e30dc638dbc88b9036811: ASoC: codecs: Add acpi_match_table for aw88399 driver (2025-07-25 13:44:29 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-merge-window for you to fetch changes up to 614d416dd8aee2675fb591c598308a901a660db8: ASoC: SOF: Intel: hda-sdw-bpt: fix SND_SOF_SOF_HDA_SDW_BPT dependencies (2025-08-06 12:02:32 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.17 This is a relatively small set of fixes and device quirks that came in during the merge window, the AMD changes adding support for ACP 7.2 systems are all just adding IDs for the devices rather than any substantial code - the actual code is the same as for prior versions of the platform. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: SOF: Intel: hda-sdw-bpt: fix SND_SOF_SOF_HDA_SDW_BPT dependencies Baojun Xu (1): ASoC: tas2781: Fix the wrong step for TLV on tas2781 Cezary Rojewski (1): ASoC: Intel: avs: Fix uninitialized pointer error in probe() Mark Brown (1): Add audio support for acp7.2 platform Muhammad Usama Anjum (1): ASoC: SOF: amd: acp-loader: Use GFP_KERNEL for DMA allocations in resume context Peter Jakubek (1): ASoC: Intel: sof_sdw: Add quirk for Alienware Area 51 (2025) 0CCC SKU Shengjiu Wang (1): ASoC: wm8962: Clear master mode when enter runtime suspend Venkata Prasad Potturu (4): ASoC: SOF: amd: Add sof audio support for acp7.2 platform ASoC: amd: ps: Add SoundWire pci and dma driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire legacy machine driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire SOF machine driver support for acp7.2 platform include/sound/tas2781-tlv.h | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 3 +++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 1 + sound/soc/amd/acp/soc_amd_sdw_common.h | 2 ++ sound/soc/amd/ps/acp63.h | 1 + sound/soc/amd/ps/pci-ps.c | 4 ++++ sound/soc/amd/ps/ps-sdw-dma.c | 5 +++++ sound/soc/codecs/wm8962.c | 11 +++++++++++ sound/soc/intel/avs/core.c | 3 ++- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ sound/soc/sof/amd/acp-loader.c | 6 +++--- sound/soc/sof/amd/acp.c | 8 ++++++++ sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp70.c | 1 + sound/soc/sof/intel/Kconfig | 3 ++- 15 files changed, 53 insertions(+), 6 deletions(-)

2 1

[PATCH 0/4] Add audio support for acp7.2 platform
by Venkata Prasad Potturu 06 Aug '25

06 Aug '25

This patch series is to add legacy and sof audio support for acp7.2 platform. Venkata Prasad Potturu (4): ASoC: SOF: amd: Add sof audio support for acp7.2 platform ASoC: amd: ps: Add SoundWire pci and dma driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire legacy machine driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire SOF machine driver support for acp7.2 platform sound/soc/amd/acp/acp-sdw-legacy-mach.c | 3 +++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 1 + sound/soc/amd/acp/soc_amd_sdw_common.h | 2 ++ sound/soc/amd/ps/acp63.h | 1 + sound/soc/amd/ps/pci-ps.c | 4 ++++ sound/soc/amd/ps/ps-sdw-dma.c | 5 +++++ sound/soc/sof/amd/acp.c | 8 ++++++++ sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp70.c | 1 + 9 files changed, 26 insertions(+) -- 2.43.0

2 5

[PATCH v1] ASoC: tas2781: Fix the wrong step for TLV on tas2781
by Baojun Xu 06 Aug '25

06 Aug '25

The step for TLV on tas2781, should be 50 (-0.5dB). Fixes: 678f38eba1f2 ("ASoC: tas2781: Add Header file for tas2781 driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index d87263e43fdb..ef9b9f19d212 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 50, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); #endif -- 2.43.0

2 1

Re: ALSAfirewire broken / Pipewire 90ms delay
by Goran Kovac 06 Aug '25

06 Aug '25

While this is mostly a non issue for playback only, recording in studio environment is unfortunately unusable. Would be really nice if this can be solved. My RME 800 can go really low with buffer size but its always +90ms.

1 0

Re: ASoC HDMI errors with sof-hda-dsp on Intel platform (no PCM for intel-hdmi-hifiX)
by Damian Tometzki 06 Aug '25

06 Aug '25

Hi all, I'm experiencing ASoC errors on an Intel-based system using the `sof-hda-dsp` driver. After boot, the kernel logs repeated errors related to HDMI outputs: [ 90.933290] snd_hda_codec_generic ehdaudio0D2: snd_soc_find_pcm_from_dai: didn't find PCM for DAI intel-hdmi-hifi1 [ 90.933295] snd_hda_codec_generic ehdaudio0D2: ASoC error (-22): at snd_soc_dai_startup() on intel-hdmi-hifi1 ... [ 91.016259] HDMI3: ASoC error (-22): at dpcm_fe_dai_startup() on HDMI3 These errors are repeated for HDMI1, HDMI2, and HDMI3. It appears that `snd_soc_find_pcm_from_dai()` fails to resolve the DAI to a PCM, and this breaks HDMI audio initialization. **System info:** - Kernel: current Linus build 2938ae2fa7cd - Distro: Fedora 42 - HDMI monitor with audio capability isn´t connected Is there a known fix or workaround? Thanks in advance, Damian On Sat, Aug 2, 2025 at 12:46 PM Damian Tometzki <damian(a)riscv-rocks.de> wrote: > > Hi all, > > I'm experiencing ASoC errors on an Intel-based system using the > `sof-hda-dsp` driver. After boot, the kernel logs repeated errors > related to HDMI outputs: > > [ 90.933290] snd_hda_codec_generic ehdaudio0D2: > snd_soc_find_pcm_from_dai: didn't find PCM for DAI intel-hdmi-hifi1 > [ 90.933295] snd_hda_codec_generic ehdaudio0D2: ASoC error (-22): at > snd_soc_dai_startup() on intel-hdmi-hifi1 > ... > [ 91.016259] HDMI3: ASoC error (-22): at dpcm_fe_dai_startup() on HDMI3 > > These errors are repeated for HDMI1, HDMI2, and HDMI3. It appears that > `snd_soc_find_pcm_from_dai()` fails to resolve the DAI to a PCM, and > this breaks HDMI audio initialization. > > **System info:** > - Kernel: current Linus build 2938ae2fa7cd > - Distro: Fedora 42 > - HDMI monitor with audio capability isn´t connected > > Is there a known fix or workaround? > Thanks in advance, > Damian

1 0

amixer: always use line buffering for (s)events
by GitHub pull_request - opened 06 Aug '25

06 Aug '25

alsa-project/alsa-utils pull request #306 was opened from elenril: Line buffering is the obviously correct mode for (s)events, but block buffering would typically be used when piping its output to another program. Request URL : https://github.com/alsa-project/alsa-utils/pull/306 Patch URL : https://github.com/alsa-project/alsa-utils/pull/306.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v1] ALSA: hda/tas2781: Support L"SmartAmpCalibrationData" to save calibrated data
by Shenghao Ding 06 Aug '25

06 Aug '25

Some devices save the calibrated data into L"CALI_DATA", and others into L"SmartAmpCalibrationData". Driver code will support both. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/hda/codecs/side-codecs/tas2781_hda.c | 47 +++++++++++++++------- sound/hda/codecs/side-codecs/tas2781_hda.h | 2 +- 2 files changed, 33 insertions(+), 16 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index 34217ce9f28e..f46d2e06c64f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -18,6 +18,8 @@ #include "tas2781_hda.h" +#define CALIBRATION_DATA_AREA_NUM 2 + const efi_guid_t tasdev_fct_efi_guid[] = { /* DELL */ EFI_GUID(0xcc92382d, 0x6337, 0x41cb, 0xa8, 0x8b, 0x8e, 0xce, 0x74, @@ -160,36 +162,51 @@ int tas2781_save_calibration(struct tas2781_hda *hda) * manufactory. */ efi_guid_t efi_guid = tasdev_fct_efi_guid[LENOVO]; - static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; + /* + * Some devices save the calibrated data into L"CALI_DATA", + * and others into L"SmartAmpCalibrationData". + */ + static efi_char16_t *efi_name[CALIBRATION_DATA_AREA_NUM] = { + L"CALI_DATA", + L"SmartAmpCalibrationData", + }; struct tasdevice_priv *p = hda->priv; struct calidata *cali_data = &p->cali_data; unsigned long total_sz = 0; unsigned int attr, size; unsigned char *data; efi_status_t status; + int i; if (hda->catlog_id < LENOVO) efi_guid = tasdev_fct_efi_guid[hda->catlog_id]; cali_data->cali_dat_sz_per_dev = 20; size = p->ndev * (cali_data->cali_dat_sz_per_dev + 1); - /* Get real size of UEFI variable */ - status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, NULL); - cali_data->total_sz = total_sz > size ? total_sz : size; - if (status == EFI_BUFFER_TOO_SMALL) { - /* Allocate data buffer of data_size bytes */ - data = p->cali_data.data = devm_kzalloc(p->dev, - p->cali_data.total_sz, GFP_KERNEL); - if (!data) { - p->cali_data.total_sz = 0; - return -ENOMEM; + for (i = 0; i < CALIBRATION_DATA_AREA_NUM; i++) { + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name[i], &efi_guid, &attr, + &total_sz, NULL); + cali_data->total_sz = total_sz > size ? total_sz : size; + if (status == EFI_BUFFER_TOO_SMALL) { + /* Allocate data buffer of data_size bytes */ + data = cali_data->data = devm_kzalloc(p->dev, + cali_data->total_sz, GFP_KERNEL); + if (!data) { + status = -ENOMEM; + continue; + } + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name[i], &efi_guid, + &attr, &cali_data->total_sz, data); } - /* Get variable contents into buffer */ - status = efi.get_variable(efi_name, &efi_guid, &attr, - &p->cali_data.total_sz, data); + /* Check whether get the calibrated data */ + if (status == EFI_SUCCESS) + break; } + if (status != EFI_SUCCESS) { - p->cali_data.total_sz = 0; + cali_data->total_sz = 0; return status; } diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.h b/sound/hda/codecs/side-codecs/tas2781_hda.h index 575a701c8dfb..66188909a0bb 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.h +++ b/sound/hda/codecs/side-codecs/tas2781_hda.h @@ -11,7 +11,7 @@ /* Flag of calibration registers address. */ #define TASDEV_UEFI_CALI_REG_ADDR_FLG BIT(7) -#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" + #define TASDEV_CALIB_N 5 /* -- 2.43.0

2 1

ALC4082: capturing S/PDIF output causes volume to jump to max - asus rog crosshair x670e hero
by GitHub issues - edited 06 Aug '25

06 Aug '25

alsa-project/alsa-ucm-conf issue #598 was edited from LuanVSO: hi, i had a problem where opening the kde plasma volume applet would make the volume on my headphone jump to the maximum. after muting the "PCM 2" channel on alsamixer i notice that the volume on the headphone would go muted instead. so i edited the ALC4080-HiFi.conf file and remove every mention of S/PDIF and rebooted my machine. after the change the problem didn't happen anymore. what do you need to fix the mapping problem? i think this happens because the "PCM 2" volume controls is redirecting to headphone output, but it doenst have a slider, so it is either max or muted: <img width="1259" height="967" alt="Image" src="https://github.com/user-attachments/assets/3b266bb7-7e1c-4a7a-b448-affabf06…" /> (also the S/PDIF aways appeared as connected even though i did not have anything connected to it, don't know if it is expected) [alsa-info.txt](https://github.com/user-attachments/files/21583734/alsa-info… <details> <summary>edited ALC4080-HiFi.conf</summary> ``` Define { SpeakerMixer "PCM" SpeakerMindex "0" SpeakerJack "Speaker - Output Jack" SpeakerPCM "hw:${CardId}" HeadphonesName "Front Headphones" HeadphonesMixer "PCM" HeadphonesMindex "1" HeadphonesJack "Headphone - Output Jack" HeadphonesPCM "hw:${CardId},1" Mic1Name "Microphone" Mic1Mixer "Mic" Mic1Mindex "0" Mic1Jack "Mic - Input Jack" Mic1PCM "hw:${CardId},2" Mic2Name "" Mic2Mixer "" Mic2Mindex "0" Mic2Jack "" Mic2PCM "" Line1Name "Line Input" Line1Mixer "Line" Line1Mindex "0" Line1Jack "Line - Input Jack" Line1PCM "hw:${CardId},1" # SpdifName "S/PDIF Output" # SpdifPCM "hw:${CardId},3" # SpdifMixer "PCM" # SpdifMindex "2" } If.speaker_ctl { Condition { Type ControlExists Control "name='Speaker Playback Switch'" } True.Define.SpeakerMixer "Speaker" } If.front_hp_ctl { Condition { Type ControlExists Control "name='Front Headphone Playback Switch'" } True.Define { HeadphonesMixer "Front Headphone" HeadphonesMindex "0" } } #If.spdif_ctl { # Condition { # Type ControlExists # Control "name='IEC958 Playback Switch'" # } # True.Define { # SpdifMixer "IEC958" # SpdifMindex "0" # } #} #If.spdif_nodev { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0db0:36e7)" # } # True.Define.SpdifName "" #} #If.spdif_dev2 { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0b05:(1996|1a5[23c])|0db0:1feb)" # } # True.Define.SpdifPCM "hw:${CardId},2" #} If.gigabyte-aorus-ultra { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0414:a014)" } True.Define { Mic1Name "Front Microphone" Mic1PCM "hw:${CardId},0" Mic2Name "Rear Microphone" Mic2Mixer "Mic" Mic2Mindex "1" Mic2Jack "name='Mic - Input Jack',index=1" Mic2PCM "hw:${CardId},1" SpdifName "" Line1Name "" } } If.asus-rog-usb { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1996)" } True.Define { Mic1Name "Front Microphone" Mic2Name "Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.asrock-taichi { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(26ce:0a0[68b])" } True.Define { Line1Name "" Mic1Name "Rear Input" Mic1PCM "hw:${CardId},0" Mic2Name "Front Microphone" Mic2Mixer "Mic" Mic2Jack "Front Mic - Input Jack" Mic2PCM "hw:${CardId},1" } } If.wrx80-sage { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1984)" } True.Define { Mic2Name "Front Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.msi-meg-unify { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0db0:82c7)" } True.Define { HeadphonesName "" SpdifName "" } } If.asus-rog-strix { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1999)" } True.Define { Line1PCM "hw:${CardId},3" HeadphonesName "" SpdifName "" } } SectionVerb { EnableSequence [ disdevall "" ] Value.TQ "HiFi" } SectionDevice."Speaker" { Comment "Speakers" EnableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} on" ] DisableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} off" ] Value { PlaybackChannels 2 PlaybackPriority 200 PlaybackPCM "${var:SpeakerPCM}" JackControl "${var:SpeakerJack}" PlaybackMixerElem "${var:SpeakerMixer},${var:SpeakerMindex}" } Variant."HiFi 5+1".Value.PlaybackChannels 6 Variant."HiFi 7+1".Value.PlaybackChannels 8 } If.headphones { Condition { Type String Empty "${var:HeadphonesName}" } False.SectionDevice."Headphones" { Comment "${var:HeadphonesName}" EnableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} on" ] DisableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} off" ] Value { PlaybackPriority 300 PlaybackPCM "${var:HeadphonesPCM}" JackControl "${var:HeadphonesJack}" PlaybackMixerElem "${var:HeadphonesMixer},${var:HeadphonesMindex}" } } } #If.spdif { # Condition { # Type String # Empty "${var:SpdifName}" # } # False.SectionDevice."SPDIF" { # Comment "${var:SpdifName}" # # EnableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} on" # ] # # DisableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} off" # ] # # Value { # PlaybackPriority 100 # PlaybackPCM "${var:SpdifPCM}" # PlaybackMixerElem "${var:SpdifMixer},${var:SpdifMindex}" # } # } #} If.line1 { Condition { Type String Empty "${var:Line1Name}" } False.SectionDevice."Line1" { Comment "${var:Line1Name}" EnableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} on" ] DisableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} off" ] Value { CapturePriority 100 CapturePCM "${var:Line1PCM}" JackControl "${var:Line1Jack}" CaptureMixerElem "${var:Line1Mixer},${var:Line1Mindex}" } } } If.mic1 { Condition { Type String Empty "${var:Mic1Name}" } False.SectionDevice."Mic1" { Comment "${var:Mic1Name}" EnableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} on" ] DisableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} off" ] Value { CapturePriority 300 CapturePCM "${var:Mic1PCM}" JackControl "${var:Mic1Jack}" CaptureMixerElem "${var:Mic1Mixer},${var:Mic1Mindex}" } } } If.mic2 { Condition { Type String Empty "${var:Mic2Name}" } False.SectionDevice."Mic2" { Comment "${var:Mic2Name}" EnableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} on" ] DisableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} off" ] Value { CapturePriority 400 CapturePCM "${var:Mic2PCM}" JackControl "${var:Mic2Jack}" CaptureMixerElem "${var:Mic2Mixer},${var:Mic2Mindex}" } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/598 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ALSA: hda: Fix the wrong register was used for DVC of TAS2770
by Baojun Xu 05 Aug '25

05 Aug '25

The wrong register was used for digital volume control of TAS2770, The definition was changed, and usage was also updated. Fixes: ab29b3460c5c ("ALSA: hda: Add TAS2770 support") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2770-tlv.h | 4 ++-- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h index c0bd495b4a07..5518b34abdde 100644 --- a/include/sound/tas2770-tlv.h +++ b/include/sound/tas2770-tlv.h @@ -14,10 +14,10 @@ #ifndef __TAS2770_TLV_H__ #define __TAS2770_TLV_H__ -#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x05) #define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) -static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, -1000, 50, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index a0b132681804..45ac5e41bd4f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -260,7 +260,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { 0, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2770_DVC_LEVEL, - 0, 0, 31, 0, tas2781_amp_getvol, + 0, 0, 200, 1, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_dvc_tlv), }; -- 2.43.0

2 1

hdajackretask.desktop missing icon
by GitHub issues - opened 05 Aug '25

05 Aug '25

alsa-project/alsa-tools issue #38 was opened from Dejweed: The hdajackretask.desktop file doesn't have the `Icon=hdajackretask`. Issue URL : https://github.com/alsa-project/alsa-tools/issues/38 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Subject: [BUG] snd_hda_intel runtime power management unsupported on MSI MPG Z390 GAMING PLUS with Realtek codec (kernel 6.8.0) causing excessive audio power draw
by Thomas van Houten 05 Aug '25

05 Aug '25

Dear Linux kernel and ALSA developers, I am reporting an issue related to runtime power management support for the onboard Realtek audio codec on my desktop motherboard, which results in continuously high audio hardware power usage and increased idle system power draw. *System Information:* * Kernel: 6.8.0-64-generic x86_64 * Distribution: Linux Mint 21.3 Virginia * Motherboard: MSI MPG Z390 GAMING PLUS (MS-7B51) * CPU: Intel Core i5-9400F * Audio Driver: snd_hda_intel * Audio Codec: Realtek (likely ALC1220 or similar) *Issue Summary:* The onboard Realtek audio codec does not enter runtime power management low-power states, causing it to remain at 100% power usage even when idle. This significantly contributes to my system's high idle power consumption (~160W total), despite low CPU and GPU draw. *Detailed Diagnostics:* 1. *PCI Audio Devices:* text |00:1f.3 Audio device [0403]: Intel Corporation Cannon Lake PCH cAVS [8086:a348] (rev 10) 03:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Navi 31 HDMI/DP Audio [1002:ab30] 07:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Navi 21/23 HDMI/DP Audio Controller [1002:ab28] | 2. *Kernel Messages related to snd_hda_intel or audio:* No logs found via|dmesg | grep -i snd_hda_intel|or|dmesg | grep -i audio|. 3. *Power management sysfs settings:* text |cat /sys/module/snd_hda_intel/parameters/power_save 10 cat /sys/class/sound/hwC*/power/runtime_status unsupported unsupported unsupported cat /sys/class/sound/hwC*/power/control auto auto auto | The|runtime_status|reports "unsupported", indicating no runtime PM support is available for these audio devices. 4. *PulseAudio and user-space processes inspection:* Stopped PulseAudio and all audio clients, with no change in power usage or device ownership of audio controls, confirming this is not caused by user-space audio applications. 5. *Powertop Reports:* text |100.0% Device Audio codec hwC2D0: Realtek | Consistently 100% usage from the Realtek audio codec, confirmed by powertop. 6. *Driver parameter set:* |snd_hda_intel|module|power_save|parameter successfully set to 10 seconds (value 10), but no effective power saving observed. *Impact:* The lack of runtime power management support for this audio codec causes continuous high power draw, increasing system idle power consumption significantly, reducing energy efficiency. *Request:* Please investigate whether this is a driver limitation, missing firmware quirk, or configuration issue with|snd_hda_intel|on the MSI MPG Z390 GAMING PLUS motherboard's Realtek codec. Guidance, workarounds, or patches to enable runtime power management would be appreciated. *Additional notes:* * Tested power management settings according to current best practices. * BIOS is up to date as of the latest MSI release. * System behavior consistent on Live USB of the same Linux Mint version—problem not due to local configuration. Thank you for your attention to this matter. I’m happy to provide additional logs or assist with testing patches. Best regards, Thomas van Houten

1 0

ucm2: Qualcomm: add Lenovo Ideapad 5 (Slim 5x / 2in1) support
by GitHub pull_request - opened 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf pull request #599 was opened from jglathe: Same layout as T14s, 2 speakers, headphone jack, DMIC Tested on Ideapad 5 2in1 14Q8X9. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/599 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/599.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALC4082: S/PDIF volume redirecting to headphone volume - asus rog crosshair x670e hero
by GitHub issues - edited 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf issue #598 was edited from LuanVSO: hi, i had a problem where opening the kde plasma volume applet would make the volume on my headphone jump to the maximum. after muting the "PCM 2" channel on alsamixer i notice that the volume on the headphone would go muted instead. so i edited the ALC4080-HiFi.conf file and remove every mention of S/PDIF and rebooted my machine. after the change the problem didn't happen anymore. what do you need to fix the mapping problem? i think this happens because the "PCM 2" volume controls is redirecting to headphone output, but it doenst have a slider, so it is either max or muted: <img width="1259" height="967" alt="Image" src="https://github.com/user-attachments/assets/3b266bb7-7e1c-4a7a-b448-affabf06…" /> (also the S/PDIF aways appeared as connected even though i did not have anything connected to it, don't know if it is expected) [alsa-info.txt](https://github.com/user-attachments/files/21583734/alsa-info… <details> <summary>edited ALC4080-HiFi.conf</summary> ``` Define { SpeakerMixer "PCM" SpeakerMindex "0" SpeakerJack "Speaker - Output Jack" SpeakerPCM "hw:${CardId}" HeadphonesName "Front Headphones" HeadphonesMixer "PCM" HeadphonesMindex "1" HeadphonesJack "Headphone - Output Jack" HeadphonesPCM "hw:${CardId},1" Mic1Name "Microphone" Mic1Mixer "Mic" Mic1Mindex "0" Mic1Jack "Mic - Input Jack" Mic1PCM "hw:${CardId},2" Mic2Name "" Mic2Mixer "" Mic2Mindex "0" Mic2Jack "" Mic2PCM "" Line1Name "Line Input" Line1Mixer "Line" Line1Mindex "0" Line1Jack "Line - Input Jack" Line1PCM "hw:${CardId},1" # SpdifName "S/PDIF Output" # SpdifPCM "hw:${CardId},3" # SpdifMixer "PCM" # SpdifMindex "2" } If.speaker_ctl { Condition { Type ControlExists Control "name='Speaker Playback Switch'" } True.Define.SpeakerMixer "Speaker" } If.front_hp_ctl { Condition { Type ControlExists Control "name='Front Headphone Playback Switch'" } True.Define { HeadphonesMixer "Front Headphone" HeadphonesMindex "0" } } #If.spdif_ctl { # Condition { # Type ControlExists # Control "name='IEC958 Playback Switch'" # } # True.Define { # SpdifMixer "IEC958" # SpdifMindex "0" # } #} #If.spdif_nodev { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0db0:36e7)" # } # True.Define.SpdifName "" #} #If.spdif_dev2 { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0b05:(1996|1a5[23c])|0db0:1feb)" # } # True.Define.SpdifPCM "hw:${CardId},2" #} If.gigabyte-aorus-ultra { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0414:a014)" } True.Define { Mic1Name "Front Microphone" Mic1PCM "hw:${CardId},0" Mic2Name "Rear Microphone" Mic2Mixer "Mic" Mic2Mindex "1" Mic2Jack "name='Mic - Input Jack',index=1" Mic2PCM "hw:${CardId},1" SpdifName "" Line1Name "" } } If.asus-rog-usb { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1996)" } True.Define { Mic1Name "Front Microphone" Mic2Name "Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.asrock-taichi { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(26ce:0a0[68b])" } True.Define { Line1Name "" Mic1Name "Rear Input" Mic1PCM "hw:${CardId},0" Mic2Name "Front Microphone" Mic2Mixer "Mic" Mic2Jack "Front Mic - Input Jack" Mic2PCM "hw:${CardId},1" } } If.wrx80-sage { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1984)" } True.Define { Mic2Name "Front Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.msi-meg-unify { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0db0:82c7)" } True.Define { HeadphonesName "" SpdifName "" } } If.asus-rog-strix { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1999)" } True.Define { Line1PCM "hw:${CardId},3" HeadphonesName "" SpdifName "" } } SectionVerb { EnableSequence [ disdevall "" ] Value.TQ "HiFi" } SectionDevice."Speaker" { Comment "Speakers" EnableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} on" ] DisableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} off" ] Value { PlaybackChannels 2 PlaybackPriority 200 PlaybackPCM "${var:SpeakerPCM}" JackControl "${var:SpeakerJack}" PlaybackMixerElem "${var:SpeakerMixer},${var:SpeakerMindex}" } Variant."HiFi 5+1".Value.PlaybackChannels 6 Variant."HiFi 7+1".Value.PlaybackChannels 8 } If.headphones { Condition { Type String Empty "${var:HeadphonesName}" } False.SectionDevice."Headphones" { Comment "${var:HeadphonesName}" EnableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} on" ] DisableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} off" ] Value { PlaybackPriority 300 PlaybackPCM "${var:HeadphonesPCM}" JackControl "${var:HeadphonesJack}" PlaybackMixerElem "${var:HeadphonesMixer},${var:HeadphonesMindex}" } } } #If.spdif { # Condition { # Type String # Empty "${var:SpdifName}" # } # False.SectionDevice."SPDIF" { # Comment "${var:SpdifName}" # # EnableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} on" # ] # # DisableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} off" # ] # # Value { # PlaybackPriority 100 # PlaybackPCM "${var:SpdifPCM}" # PlaybackMixerElem "${var:SpdifMixer},${var:SpdifMindex}" # } # } #} If.line1 { Condition { Type String Empty "${var:Line1Name}" } False.SectionDevice."Line1" { Comment "${var:Line1Name}" EnableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} on" ] DisableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} off" ] Value { CapturePriority 100 CapturePCM "${var:Line1PCM}" JackControl "${var:Line1Jack}" CaptureMixerElem "${var:Line1Mixer},${var:Line1Mindex}" } } } If.mic1 { Condition { Type String Empty "${var:Mic1Name}" } False.SectionDevice."Mic1" { Comment "${var:Mic1Name}" EnableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} on" ] DisableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} off" ] Value { CapturePriority 300 CapturePCM "${var:Mic1PCM}" JackControl "${var:Mic1Jack}" CaptureMixerElem "${var:Mic1Mixer},${var:Mic1Mindex}" } } } If.mic2 { Condition { Type String Empty "${var:Mic2Name}" } False.SectionDevice."Mic2" { Comment "${var:Mic2Name}" EnableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} on" ] DisableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} off" ] Value { CapturePriority 400 CapturePCM "${var:Mic2PCM}" JackControl "${var:Mic2Jack}" CaptureMixerElem "${var:Mic2Mixer},${var:Mic2Mindex}" } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/598 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALC4082: S/PDIF volume redirecting to headphone volume - asus rog crosshair x670e hero
by GitHub issues - opened 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf issue #598 was opened from LuanVSO: hi, i had a problem where opening the kde plasma volume applet would make the volume on my headphone jump to the maximum. after muting the "PCM 2" channel on alsamixer i notice that the volume on the headphone would go muted instead. so i edited the ALC4080-HiFi.conf file and remove every mention of S/PDIF and rebooted my machine. after the change the problem didn't happen anymore. what do you need to fix the mapping problem? i think this happens because the "PCM 2" volume controls is redirecting to headphone output, but it doenst have a slider, so it is either max or muted: <img width="1259" height="967" alt="Image" src="https://github.com/user-attachments/assets/3b266bb7-7e1c-4a7a-b448-affabf06…" /> (also the S/PDIF aways appeared as connected even though i did not have anything connected to it, don't know if it is expected) [alsa-info.sh](https://paste.centos.org/view/c6545d44) <details> <summary>edited ALC4080-HiFi.conf</summary> ``` Define { SpeakerMixer "PCM" SpeakerMindex "0" SpeakerJack "Speaker - Output Jack" SpeakerPCM "hw:${CardId}" HeadphonesName "Front Headphones" HeadphonesMixer "PCM" HeadphonesMindex "1" HeadphonesJack "Headphone - Output Jack" HeadphonesPCM "hw:${CardId},1" Mic1Name "Microphone" Mic1Mixer "Mic" Mic1Mindex "0" Mic1Jack "Mic - Input Jack" Mic1PCM "hw:${CardId},2" Mic2Name "" Mic2Mixer "" Mic2Mindex "0" Mic2Jack "" Mic2PCM "" Line1Name "Line Input" Line1Mixer "Line" Line1Mindex "0" Line1Jack "Line - Input Jack" Line1PCM "hw:${CardId},1" # SpdifName "S/PDIF Output" # SpdifPCM "hw:${CardId},3" # SpdifMixer "PCM" # SpdifMindex "2" } If.speaker_ctl { Condition { Type ControlExists Control "name='Speaker Playback Switch'" } True.Define.SpeakerMixer "Speaker" } If.front_hp_ctl { Condition { Type ControlExists Control "name='Front Headphone Playback Switch'" } True.Define { HeadphonesMixer "Front Headphone" HeadphonesMindex "0" } } #If.spdif_ctl { # Condition { # Type ControlExists # Control "name='IEC958 Playback Switch'" # } # True.Define { # SpdifMixer "IEC958" # SpdifMindex "0" # } #} #If.spdif_nodev { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0db0:36e7)" # } # True.Define.SpdifName "" #} #If.spdif_dev2 { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0b05:(1996|1a5[23c])|0db0:1feb)" # } # True.Define.SpdifPCM "hw:${CardId},2" #} If.gigabyte-aorus-ultra { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0414:a014)" } True.Define { Mic1Name "Front Microphone" Mic1PCM "hw:${CardId},0" Mic2Name "Rear Microphone" Mic2Mixer "Mic" Mic2Mindex "1" Mic2Jack "name='Mic - Input Jack',index=1" Mic2PCM "hw:${CardId},1" SpdifName "" Line1Name "" } } If.asus-rog-usb { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1996)" } True.Define { Mic1Name "Front Microphone" Mic2Name "Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.asrock-taichi { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(26ce:0a0[68b])" } True.Define { Line1Name "" Mic1Name "Rear Input" Mic1PCM "hw:${CardId},0" Mic2Name "Front Microphone" Mic2Mixer "Mic" Mic2Jack "Front Mic - Input Jack" Mic2PCM "hw:${CardId},1" } } If.wrx80-sage { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1984)" } True.Define { Mic2Name "Front Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.msi-meg-unify { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0db0:82c7)" } True.Define { HeadphonesName "" SpdifName "" } } If.asus-rog-strix { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1999)" } True.Define { Line1PCM "hw:${CardId},3" HeadphonesName "" SpdifName "" } } SectionVerb { EnableSequence [ disdevall "" ] Value.TQ "HiFi" } SectionDevice."Speaker" { Comment "Speakers" EnableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} on" ] DisableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} off" ] Value { PlaybackChannels 2 PlaybackPriority 200 PlaybackPCM "${var:SpeakerPCM}" JackControl "${var:SpeakerJack}" PlaybackMixerElem "${var:SpeakerMixer},${var:SpeakerMindex}" } Variant."HiFi 5+1".Value.PlaybackChannels 6 Variant."HiFi 7+1".Value.PlaybackChannels 8 } If.headphones { Condition { Type String Empty "${var:HeadphonesName}" } False.SectionDevice."Headphones" { Comment "${var:HeadphonesName}" EnableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} on" ] DisableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} off" ] Value { PlaybackPriority 300 PlaybackPCM "${var:HeadphonesPCM}" JackControl "${var:HeadphonesJack}" PlaybackMixerElem "${var:HeadphonesMixer},${var:HeadphonesMindex}" } } } #If.spdif { # Condition { # Type String # Empty "${var:SpdifName}" # } # False.SectionDevice."SPDIF" { # Comment "${var:SpdifName}" # # EnableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} on" # ] # # DisableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} off" # ] # # Value { # PlaybackPriority 100 # PlaybackPCM "${var:SpdifPCM}" # PlaybackMixerElem "${var:SpdifMixer},${var:SpdifMindex}" # } # } #} If.line1 { Condition { Type String Empty "${var:Line1Name}" } False.SectionDevice."Line1" { Comment "${var:Line1Name}" EnableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} on" ] DisableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} off" ] Value { CapturePriority 100 CapturePCM "${var:Line1PCM}" JackControl "${var:Line1Jack}" CaptureMixerElem "${var:Line1Mixer},${var:Line1Mindex}" } } } If.mic1 { Condition { Type String Empty "${var:Mic1Name}" } False.SectionDevice."Mic1" { Comment "${var:Mic1Name}" EnableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} on" ] DisableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} off" ] Value { CapturePriority 300 CapturePCM "${var:Mic1PCM}" JackControl "${var:Mic1Jack}" CaptureMixerElem "${var:Mic1Mixer},${var:Mic1Mindex}" } } } If.mic2 { Condition { Type String Empty "${var:Mic2Name}" } False.SectionDevice."Mic2" { Comment "${var:Mic2Name}" EnableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} on" ] DisableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} off" ] Value { CapturePriority 400 CapturePCM "${var:Mic2PCM}" JackControl "${var:Mic2Jack}" CaptureMixerElem "${var:Mic2Mixer},${var:Mic2Mindex}" } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/598 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Regression: Microphone broken on Intel Ice Lake with alsa-ucm-conf 1.2.10-150600.3.5.1 (openSUSE Leap 15.6)
by Pablo Scatizza 04 Aug '25

04 Aug '25

Dear ALSA developers, I would like to report a regression affecting the microphone on my laptop using Intel's Ice Lake-LP audio controller with the `sof-hda-dsp` driver. After upgrading `alsa-ucm-conf` to version 1.2.10-150600.3.5.1 (as distributed in openSUSE Leap 15.6), the internal microphone stops working. No capture is possible via `arecord` or any application. The device still shows up in `arecord -l`, but it does not receive audio input. Rolling back to the previous version (1.2.10-150600.1.2) using: sudo zypper install --oldpackage alsa-ucm-conf-1.2.10-150600.1.2.noarch restores full microphone functionality. **System information:** - Distribution: openSUSE Leap 15.6 - Kernel: 6.4.0-150600.23.53-default - Audio device: Intel Corporation Ice Lake-LP Smart Sound Technology Audio Controller (rev 30) - ALSA driver: sof-hda-dsp **Working `arecord -l` output with the old version:** card 0: sofhdadsp [sof-hda-dsp], device 0: HDA Analog ( *) [] card 0: sofhdadsp [sof-hda-dsp], device 6: DMIC (*) [] card 0: sofhdadsp [sof-hda-dsp], device 7: DMIC16kHz (*) [] Let me know if I can provide logs, diffs between UCM files, or further testing. I'd be happy to help. Best regards, Pablo Scatizza Neuquén, Argentina

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Qualcomm: Add QCS8275-IQ8-EVK HiFi config
by GitHub pull_request - opened 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was opened from mohsRafi: Add UCM2 configs for the Qualcomm QCS9075-IQ8-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 1/2] ALSA: hda/realtek: Accelerate the front MIC charging and discharging of the 897 codec
by wangdich9700＠163.com 04 Aug '25

04 Aug '25

From: wangdicheng <wangdicheng(a)kylinos.cn> The front MIC capacitance of the Inspur board is too large, resulting in a longer charging and discharging time required, which in turn prevents recording for 5 seconds before startup. Therefore, software is used to increase the delay by 2 seconds to avoid this issue. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 877137cb09ac..88675eae9447 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1001,6 +1001,14 @@ static int alc_resume(struct hda_codec *codec) if (!spec->no_depop_delay) msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); + + if (codec->core.subsystem_id == 0x10ec1304) { + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04); + msleep(50); + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44); + msleep(2000); + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80); + } snd_hda_regmap_sync(codec); hda_call_check_power_status(codec, 0x01); return 0; -- 2.25.1

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[PATCH REQUEST] ALSA: hda/tas2781: Add support for ASUS VivoBook 16 Flip TP3607SA
by Ludd ite 02 Aug '25

02 Aug '25

Summary ======= The ASUS VivoBook 16 Flip TP3607SA laptop contains TI TAS2781 smart amplifiers but they are not supported by the current snd_hda_scodec_tas2781_i2c driver. The driver fails to initialize with "Platform not supported" error, preventing internal speakers from working. Hardware Information ==================== Laptop Model: ASUS VivoBook 16 Flip TP3607SA_TP3607SA BIOS Version: 1.0 PCI Audio Device: Intel Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem Vendor: 0x1043 (ASUS) Subsystem Device: 0x3e30 Audio Codec: Realtek ALC294 (HDA:10ec0294,10433e30,00100004) Smart Amplifier: TI TAS2781 on I2C bus (ACPI ID: TIAS2781:00) System Information ================== OS: Ubuntu 25.04 (development) Kernel: 6.14.0-1007-realtime Audio System: PipeWire with SOF (Sound Open Firmware) SOF Firmware: intel/sof-ipc4/lnl/sof-lnl.ri (version 2.12.0.1) Topology: intel/sof-ipc4-tplg/sof-hda-generic-2ch.tplg Current Behavior ================ 1. Audio Detection: SOF firmware loads successfully and creates speaker sink in PipeWire 2. Audio Pipeline: Complete audio pipeline works (volume controls, routing, mixing) 3. TAS2781 Driver: Fails to bind with the following errors: tas2781-hda i2c-TIAS2781:00: Failed to get Speaker id. tas2781-hda i2c-TIAS2781:00: read acpi error, ret: -2 tas2781-hda i2c-TIAS2781:00: error -ENOENT: Platform not supported tas2781-hda i2c-TIAS2781:00: probe with driver tas2781-hda failed with error -2 4. Result: No sound from internal speakers (Bluetooth and HDMI audio work fine) Technical Analysis ================== - I2C Device Present: i2c-TIAS2781:00 device exists and is detected - ACPI Information: Device has ACPI ID TIAS2781: but driver cannot read required configuration - Volume Control: Audio shows active volume bars in pavucontrol, indicating software pipeline works - Driver Status: TAS2781 driver loads but doesn't bind to the device due to platform whitelist Evidence of TAS2781 Hardware ============================= I2C device exists: $ ls /sys/bus/i2c/devices/i2c-TIAS2781:00/ firmware_node modalias name power subsystem Device modalias: $ cat /sys/bus/i2c/devices/i2c-TIAS2781:00/modalias acpi:TIAS2781: PipeWire detects speaker sink: $ pactl list sinks short | grep Speaker 60 alsa_output.pci-0000_00_1f.3-platform-skl_hda_dsp_generic.HiFi__Speaker__sink Expected Behavior ================= The TAS2781 driver should recognize the ASUS VivoBook 16 Flip TP3607SA (subsystem ID 1043:3e30) as a supported platform and successfully initialize the smart amplifiers, enabling internal speaker audio output. Workarounds Attempted ===================== 1. Manual I2C device creation: Created tas2781 devices at addresses 0x38-0x3b, but they fail with communication timeouts (error -121) 2. Driver parameters: Tried various modprobe options, but the driver doesn't expose platform override parameters 3. GPIO manipulation: Attempted to enable GPIO pins that might control amplifier power, without success 4. Kernel versions: Tested multiple 6.14.x kernels including recent versions with November 26, 2024 ASUS TAS2781 improvements Request ======= Please add support for ASUS VivoBook 16 Flip TP3607SA (subsystem ID 1043:3e30) to the TAS2781 driver's supported platform list. This appears to be a matter of adding the device ID to the driver's whitelist and possibly providing the appropriate ACPI configuration parsing. Additional Notes ================ - This is a recent laptop model (2024) with Intel Lunar Lake platform - Other ASUS laptops may have similar TAS2781 configurations - The November 26, 2024 commit "ALSA: hda/tas2781: Add speaker id check for ASUS projects" suggests ongoing work on ASUS TAS2781 support - Hardware detection is working perfectly through SOF firmware Contact Information =================== Available for testing patches or providing additional debugging information. Attachments =========== - Full dmesg output available upon request - ACPI tables dump available if needed - Detailed hardware information via sudo lshw available

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conf/pistachio: fix syntax
by GitHub pull_request - opened 01 Aug '25

01 Aug '25

alsa-project/alsa-lib pull request #470 was opened from blino: It was missing closing brackets since its introduction. Fixes: 4dfa8f08fb83 ("conf/cards: add support for pistachio-card.") Request URL : https://github.com/alsa-project/alsa-lib/pull/470 Patch URL : https://github.com/alsa-project/alsa-lib/pull/470.patch Repository URL: https://github.com/alsa-project/alsa-lib

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retire_capture_urb: Corrected urb data len
by Sean Anderson 01 Aug '25

01 Aug '25

Hi all, I have a "FiiO DigiHug USB Audio" sound card (1852:7022) [3]. I have had no problems with the audio, but I did notice a large number of message like retire_capture_urb: 4992 callbacks suppressed in my dmesg [1]. This is caused by the "Corrected urb data len." warning. The patch adding this warning [2] makes it seem like this warning should be an uncommon occurance. However, based on the number of suppressed callbacks, this seems to be happening at a rate of around 500 Hz. Is this buggy hardware? Or is this a bug in the driver? Does there need to be a quirk? Or perhaps the warning above should be a debug instead? I'm using 5.15.0-46-generic. I know, I know; this is a vendor kernel. However, I went through the commits for sound/usb/pcm.c and didn't see any major differences. I also previously saw this issue on 5.4, so this probably isn't a regression. --Sean [1] I would have liked to post the actual message, but I was unable to figure out how to disable ratelimiting. I tried setting /proc/sys/kernel/printk_ratelimit to 0, /proc/sys/kernel/printk_ratelimit_burst to 5000, and /proc/sys/kernel/printk to 8. However, nothing seemed to have any effect (except that the callbacks suppressed messages were disabled). I find this very strange, given that the callbacks suppressed message uses KERN_WARN, and so does dev_warn_ratelimited. [2] https://lore.kernel.org/all/4B28A659.6070303@ladisch.de/T/ [3] Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x1852 GYROCOM C&C Co., LTD idProduct 0x7022 bcdDevice 0.01 iManufacturer 1 FiiO iProduct 2 DigiHug USB Audio iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0182 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 58 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0012 1x 18 bytes bInterval 32 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 3 FiiO USB DAC-E10 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x003e bInCollection 2 baInterfaceNr(0) 2 baInterfaceNr(1) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 9 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bSourceID 16 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 16 bSourceID 9 bControlSize 1 bmaControls(0) 0x01 Mute Control bmaControls(1) 0x02 Volume Control bmaControls(2) 0x02 Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x2001 IEC1937_AC-3 AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 3 (FORMAT_TYPE_III) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0000 (Bus Powered)

2 8

pcm: add a loop to snd_pcm_avail_delay() to avoid bogus delay values
by GitHub pull_request - opened 31 Jul '25

31 Jul '25

alsa-project/alsa-lib pull request #469 was opened from kv2019i: snd_pcm_avail_delay() is expected to report avail and delay values in atomic fashion. However the function does two separate syscalls and it cannot guarantee the avail value is the same as was used to calculate the delay. This is a problem as the reported delay is always relative to avail frames value. If application (like e.g. alsa_conformance_test) uses snd_pcm_avail_delay() to estimate the effective play position, it can observe bogus delay values (and effective play position going backwards) if snd_pcm_avail_delay() is called during a DMA burst where hw_ptr moves quickly. This commit adds a loop similar to that used in snd_pcm_hw_htimestamp() to wait until we get a stable avail reading, and only then extract the delay. This will avoid bogus values if function is called during DMA bursts. Closes: https://github.com/alsa-project/alsa-lib/issues/468 Request URL : https://github.com/alsa-project/alsa-lib/pull/469 Patch URL : https://github.com/alsa-project/alsa-lib/pull/469.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_pcm_avail_delay() reporting inaccurate delay values
by GitHub issues - opened 31 Jul '25

31 Jul '25

alsa-project/alsa-lib issue #468 was opened from kv2019i: The alsa_conformance_test tool (https://chromium.googlesource.com/chromiumos/platform/audiotest/+/HEAD/alsa…) has recently added support to use snd_pcm_avail_delay() to estimate the actual playback position. This has proven a very useful tool to test ALSA drivers on hardware (like that support by https://github.com/thesofproject/ ) where there is significant buffering in hardware (and hw_ptr can move in bursts). When testing with high sampling rates, an issue was found where the estimated playback position seemed to go backwards in time, as observed by alsa_conformance_test. This was observed on multiple Intel platform using SOF firmware for DSP. Debugging the kernel driver delay() implementation showed, that driver never reported incorrect delay values, but somehow alsa_conformance_test still observed these. Example log: ``` alsa_conformance_test -P hw:sofnocodec,2 -d 1 -r 96000 -B 480 -c 2 -f S32_LE -p 480 --avail-delay --debug ... TIME_DIFF(s) HW_LEVEL PLAYED DELAY DIFF RATE [.. snip ] 0.000999950 960 3324 1476 96 96004.800240 0.000995762 864 3420 1380 96 96408.579560 0.001024820 672 3516 1284 96 93674.986827 0.000987638 621 3613 1187 97 98214.122988 [Notice] frame_diff -1 >= 4096. 0.000049322 576 3612 1188 -1 -20274.928024 # KV: the delay value of 1187 is never reported by driver and it causes the playback position to go backwards in time despite driver reporting the accurate delay correctly 0.000992228 480 3708 1092 96 96751.956204 0.000972403 864 3804 1476 96 98724.500027 0.000998287 768 3900 1380 96 96164.730183 0.000979150 768 3996 1284 96 98044.222029 0.000999349 672 4092 1188 96 96062.536711 0.001018172 480 4188 1092 96 94286.623478 0.000977025 960 4284 1476 96 98257.465264 0.001022131 768 4380 1380 96 93921.424945 0.000977795 768 4476 1284 96 98180.088873 0.001019073 601 4573 1187 97 95184.545170 [Notice] frame_diff -1 >= 4096. 0.000049433 576 4572 1188 -1 -20229.401412 0.000953494 480 4668 1092 96 100682.332558 0.000981225 960 4764 1476 96 97836.887564 0.001025294 768 4860 1380 96 93631.680279 ``` When instrumenting alsa-lib snd_pcm_avail_delay() it became apparent the function is calling in middle of a DMA burst (when hw_ptr is moving fast ahead and the "avail" number is changing fast). alsa-lib does two syscalls to read avail and delay, the two values are not in sync and we end up reporting a bogus value. The error is not a big one, but this is enough to fail alsa_conformance_test test plans and push the calculcated standard rate error above acceptance criteria. This could also confuse media players who use this for video lip-sync, although in theory applications should handle this (a 0...3 frame error in delay reporting is not going to make a difference unless negative value breaks the application logic like happens here with alsa_conformance_test). We can observe the error especially with higher rates (96, 176.4 and 192kHz). This is harder to hit with the more typical 48khz use-cases, and thus maybe has not been hit before. Versions tested: 1.12.24, 1.12.11 Issue URL : https://github.com/alsa-project/alsa-lib/issues/468 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ucm: use close_range on _GNU_SOURCE
by Jochen Sprickerhof 30 Jul '25

30 Jul '25

--- src/ucm/ucm_exec.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/src/ucm/ucm_exec.c b/src/ucm/ucm_exec.c index 276cf592..b79a84ae 100644 --- a/src/ucm/ucm_exec.c +++ b/src/ucm/ucm_exec.c @@ -254,8 +254,12 @@ int uc_mgr_exec(const char *prog) close(f); +#if defined(_GNU_SOURCE) + close_range(3, maxfd, 0); +#else for (f = 3; f < maxfd; f++) close(f); +#endif /* install default handlers for the forked process */ signal(SIGINT, SIG_DFL); -- 2.50.0

1 0

[PATCH v2] ALSA: hda: Fix the wrong register was used for DVC of TAS2770
by Baojun Xu 29 Jul '25

29 Jul '25

The wrong register was used for digital volume control of TAS2770, The definition was changed, and usage was also updated. Fixes: ab29b3460c5c ("ALSA: hda: Add TAS2770 support") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2 - Correct number from -1000 to -10000 for DVC. --- include/sound/tas2770-tlv.h | 4 ++-- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h index c0bd495b4a07..c7380925417a 100644 --- a/include/sound/tas2770-tlv.h +++ b/include/sound/tas2770-tlv.h @@ -14,10 +14,10 @@ #ifndef __TAS2770_TLV_H__ #define __TAS2770_TLV_H__ -#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x05) #define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) -static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, -10000, 50, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index a0b132681804..45ac5e41bd4f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -260,7 +260,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { 0, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2770_DVC_LEVEL, - 0, 0, 31, 0, tas2781_amp_getvol, + 0, 0, 200, 1, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_dvc_tlv), }; -- 2.43.0

1 0

[PATCH] ASoC: amd: acp: Adjust pdm dmic gain using module parameter
by Venkata Prasad Potturu 29 Jul '25

29 Jul '25

Adjust pdm dimc gain value using module param. In case of regressions for any users that the new pdm_gain value is too high and for additional debugging, introduce a module parameter that would let them configure it. This parameter should be removed in the future: * If it's determined that the parameter is not needed, just hardcode the correct value as before * If users do end up using it to debug and report different values we should introduce a config knob that can have policy set by ucm. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/acp-legacy-common.c | 3 ++- sound/soc/amd/acp/acp-pdm.c | 3 ++- sound/soc/amd/acp/amd.h | 6 +++++- 3 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 57982d057c3a..dd804fb95790 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -173,7 +173,8 @@ static void set_acp_pdm_clk(struct snd_pcm_substream *substream, /* Enable default ACP PDM clk */ writel(PDM_CLK_FREQ_MASK, chip->base + ACP_WOV_CLK_CTRL); pdm_ctrl = readl(chip->base + ACP_WOV_MISC_CTRL); - pdm_ctrl |= PDM_MISC_CTRL_MASK; + pdm_ctrl &= ~ACP_WOV_GAIN_CONTROL; + pdm_ctrl |= FIELD_PREP(ACP_WOV_GAIN_CONTROL, clamp(pdm_gain, 0, 3)); writel(pdm_ctrl, chip->base + ACP_WOV_MISC_CTRL); set_acp_pdm_ring_buffer(substream, dai); } diff --git a/sound/soc/amd/acp/acp-pdm.c b/sound/soc/amd/acp/acp-pdm.c index 1bfc34c2aa53..ffb622a7a69a 100644 --- a/sound/soc/amd/acp/acp-pdm.c +++ b/sound/soc/amd/acp/acp-pdm.c @@ -38,7 +38,8 @@ static int acp_dmic_prepare(struct snd_pcm_substream *substream, /* Enable default DMIC clk */ writel(PDM_CLK_FREQ_MASK, chip->base + ACP_WOV_CLK_CTRL); dmic_ctrl = readl(chip->base + ACP_WOV_MISC_CTRL); - dmic_ctrl |= PDM_MISC_CTRL_MASK; + dmic_ctrl &= ~ACP_WOV_GAIN_CONTROL; + dmic_ctrl |= FIELD_PREP(ACP_WOV_GAIN_CONTROL, clamp(pdm_gain, 0, 3)); writel(dmic_ctrl, chip->base + ACP_WOV_MISC_CTRL); period_bytes = frames_to_bytes(substream->runtime, diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index cb8d97122f95..f2567e06ccd3 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -130,7 +130,7 @@ #define PDM_DMA_INTR_MASK 0x10000 #define PDM_DEC_64 0x2 #define PDM_CLK_FREQ_MASK 0x07 -#define PDM_MISC_CTRL_MASK 0x10 +#define ACP_WOV_GAIN_CONTROL GENMASK(4, 3) #define PDM_ENABLE 0x01 #define PDM_DISABLE 0x00 #define DMA_EN_MASK 0x02 @@ -138,6 +138,10 @@ #define PDM_TIMEOUT 1000 #define ACP_REGION2_OFFSET 0x02000000 +static int pdm_gain = 3; +module_param(pdm_gain, int, 0644); +MODULE_PARM_DESC(pdm_gain, "Gain control (0-3)"); + struct acp_chip_info { char *name; /* Platform name */ struct resource *res; -- 2.43.0

4 4

Support ohos build
by GitHub pull_request - opened 29 Jul '25

29 Jul '25

alsa-project/alsa-lib pull request #467 was opened from wyjstrong: https://github.com/alsa-project/alsa-lib/issues/466 Request URL : https://github.com/alsa-project/alsa-lib/pull/467 Patch URL : https://github.com/alsa-project/alsa-lib/pull/467.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Support ohos build
by GitHub issues - opened 29 Jul '25

29 Jul '25

alsa-project/alsa-lib issue #466 was opened from wyjstrong: Hello, ALSA is an important audio architecture in the Linux system. With it, developers can call Linux system interfaces in their programs to control and manage the underlying operating system. Thanks to its combination of high efficiency and ease of use, ALSA is also suitable for embedded devices, which highly aligns with OpenHarmony's philosophy of supporting diverse devices — currently, it has been widely used in the field of mobile application development for OpenHarmony. We will submit the adaptation code via a Pull Request (PR) as soon as possible. This issue is created to conduct in-depth discussions with the community on the integration of the adaptation code. We sincerely welcome your valuable suggestions. Thank you. Issue URL : https://github.com/alsa-project/alsa-lib/issues/466 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC updates for v6.17-2
by Mark Brown 28 Jul '25

28 Jul '25

The following changes since commit 347e9f5043c89695b01e66b3ed111755afcf1911: Linux 6.16-rc6 (2025-07-13 14:25:58 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.17-2 for you to fetch changes up to e95122a32e777309412e30dc638dbc88b9036811: ASoC: codecs: Add acpi_match_table for aw88399 driver (2025-07-25 13:44:29 +0100) ---------------------------------------------------------------- ASoC: More updates for v6.17 A few more updates, mostly fixes and device IDs plus some small enhancements for the FSL xcvr driver. ---------------------------------------------------------------- Ai Chao (3): ASoC: renesas: Use helper function for_each_child_of_node_scoped() ASoC: meson: Use helper function for_each_child_of_node_scoped() ASoC: imx-card: Use helper function for_each_child_of_node_scoped() Alexander Stein (1): ASoC: codec: tlv320aic32x4: Fix reset GPIO check Arnd Bergmann (2): ASoC: ops: dynamically allocate struct snd_ctl_elem_value ASoC: SDCA: fix HID dependency Balamurugan C (3): ASoC: Intel: soc-acpi: Add entry for sof_es8336 in PTL match table. ASoC: Intel: soc-acpi: Add entry for HDMI_In capture support in PTL match table ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for PTL. Bard Liao (2): ASoC: sdw_utils: add component_name string to dai_info ASoC: sdw_utils: generate combined spk components string Bram Vlerick (2): ASoC: tas571x: add support for tas5753 ASoC: dt-bindings: tas57xx: add tas5753 compatibility Brian Masney (6): ASoC: codecs: da7219: convert from round_rate() to determine_rate() ASoC: codecs: rt5682: convert from round_rate() to determine_rate() ASoC: codecs: rt5682s: convert from round_rate() to determine_rate() ASoC: qcom: qdsp6: q6dsp-lpass-clocks: convert from round_rate() to determine_rate() ASoC: stm: stm32_i2s: convert from round_rate() to determine_rate() ASoC: stm: stm32_sai_sub: convert from round_rate() to determine_rate() Cezary Rojewski (2): ASoC: codecs: rt5640: Drop dummy register names ASoC: Intel: avs: Add rt5640 machine board Chancel Liu (1): ASoC: imx-card: Add WM8524 support Charles Keepax (23): HID: core: Add bus define for SoundWire bus MAINTAINERS: Add SDCA maintainers entry ASoC: SDCA: Add missing default in switch in entity_pde_event() ASoC: SDCA: Fixup some kernel doc errors ASoC: SDCA: Minor selected/detected mode control fixups ASoC: SDCA: Add flag for unused IRQs ASoC: SDCA: Add some initial IRQ handlers ASoC: SDCA: Kconfig/Makefile fixups ASoC: SDCA: Pull HID and IRQ into the primary SDCA module ASoC: SDCA: Fix off by one error in IRQ bound check ASoC: SDCA: Avoid use of uninitialised local name variable ASoC: SDCA: Update memory allocations to zero initialise ASoC: SDCA: Allow read-only controls to be deferrable ASoC: SDCA: Remove overly chatty input pin list warning ASoC: SDCA: Move SDCA search functions and export ASoC: soc-dai: Add private data to snd_soc_dai ASoC: SDCA: Add helper to add DAI constraints ASoC: SDCA: Add a helper to get the SoundWire port number ASoC: SDCA: Add hw_params() helper function ASoC: SDCA: Shrink detected_mode_handler() stack frame ASoC: SDCA: Fix implicit cast from le16 ASoC: SDCA: Fix some holes in the regmap readable/writeable helpers ASoC: SDCA: Add support for -cn- value properties Chen Ni (1): ASoC: tegra: AHUB: Remove unneeded semicolon Chen-Yu Tsai (9): ASoC: dt-bindings: Convert MT8173 AFE binding to dt-schema ASoC: dt-bindings: mt8173-afe-pcm: Add power domain ASoC: dt-bindings: mt8173-afe-pcm: Allow specifying reserved memory region ASoC: dt-bindings: mt8186-afe-pcm: Allow specifying reserved memory region ASoC: dt-bindings: mt8192-afe-pcm: Allow specifying reserved memory region ASoC: mediatek: use reserved memory or enable buffer pre-allocation ASoC: mediatek: mt8183-afe-pcm: Support >32 bit DMA addresses ASoC: mediatek: mt8173-afe-pcm: use local `dev` pointer in driver callbacks ASoC: mediatek: mt8183-afe-pcm: use local `dev` pointer in driver callbacks ChiYuan Huang (2): ASoC: dt-bindings: rt9123: Append RTQ9124 description ASoC: codecs: Add support for Richtek RTQ9124 Frank Li (1): ASoC: dt-bindings: covert mxs-audio-sgtl5000.txt to yaml format Geert Uytterhoeven (2): ASoC: renesas: msiof: Convert to <linux/spi/sh_msiof.h> ASoC: img: Imagination Technologies sound should depend on MIPS Jyri Sarha (3): ASoC: SOF: ipc4: Add sof_ipc4_pipeline_state_str() for debugging ASoC: SOF: ipc4-pcm: Pipe instances to dev_dbg in multi_pipeline_state() ASoC: SOF: ipc4-pcm: Harmonize sof_ipc4_set_pipeline_state() dbg print Krzysztof Kozlowski (20): ASoC: codecs: Constify regmap configuration static variables ASoC: fsl: Constify reg_default array ASoC: codecs: wcd9335: Drop unused sido_input_src field ASoC: codecs: wcd934x: Drop unused num_rx_port/num_tx_port fields ASoC: codecs: wcd937x: Drop unused 'struct wcd937x_priv' fields ASoC: codecs: wcd938x: Drop unused 'struct wcd938x_priv' fields ASoC: codecs: wcd938x: Drop unused variant field ASoC: codecs: wcd939x: Drop unused 'struct wcd939x_priv' fields ASoC: codecs: wcd937x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd938x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd939x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd939x: Simplify return from devm_gpiod_get() error ASoC: dt-bindings: qcom,wcd939x: Document missing VDD_PX supply ASoC: codecs: wcd939x: Add VDD_PX supply ASoC: codecs: lpass: Drop unused AIF_INVALID first DAI identifier ASoC: codecs: wcd937x: Use simple defines for chipid register value ASoC: codecs: wcd938x: Use simple defines for chipid register value ASoC: codecs: wcd939x: Use simple defines for chipid register value ASoC: codecs: wcd939x: Add defines for major/minor version decoding ASoC: dt-bindings: qcom,lpass-va-macro: Define clock-names in top-level Kuninori Morimoto (62): ASoC: soc-core: save ID if param was set in fmt_single_name() ASoC: remove component->id ASoC: hdac_hdmi: remove hdac_hdmi_jack[_port]_init() ASoC: soc-dai: tidyup return value of snd_soc_xlate_tdm_slot_mask() ASoC: soc-ops-test: dynamically allocate struct snd_ctl_elem_value ASoC: ti: Standardize ASoC menu ASoC: adi: Standardize ASoC menu ASoC: amd: Standardize ASoC menu ASoC: bcm: Standardize ASoC menu ASoC: dwc: Standardize ASoC menu ASoC: fsl: Standardize ASoC menu ASoC: img: Standardize ASoC menu ASoC: mxs: Standardize ASoC menu ASoC: pxa: Standardize ASoC menu ASoC: sti: Standardize ASoC menu ASoC: stm: Standardize ASoC menu ASoC: sof: Standardize ASoC menu ASoC: au1x: Standardize ASoC menu ASoC: sdca: Standardize ASoC menu ASoC: sprd: Standardize ASoC menu ASoC: qcom: Standardize ASoC menu ASoC: meson: Standardize ASoC menu ASoC: apple: Standardize ASoC menu ASoC: atmel: Standardize ASoC menu ASoC: intel: Standardize ASoC menu ASoC: spear: Standardize ASoC menu ASoC: sunxi: Standardize ASoC menu ASoC: tegra: Standardize ASoC menu ASoC: ux500: Standardize ASoC menu ASoC: cirrus: Standardize ASoC menu ASoC: google: Standardize ASoC menu ASoC: jz4740: Standardize ASoC menu ASoC: xilinx: Standardize ASoC menu ASoC: xtensa: Standardize ASoC menu ASoC: renesas: Standardize ASoC menu ASoC: generic: Standardize ASoC menu ASoC: samsung: Standardize ASoC menu ASoC: kirkwood: Standardize ASoC menu ASoC: loongson: Standardize ASoC menu ASoC: mediatek: Standardize ASoC menu ASoC: rockchip: Standardize ASoC menu ASoC: starfive: Standardize ASoC menu ASoC: uniphier: Standardize ASoC menu ASoC: hisilicon: Standardize ASoC menu ASoC: soc-dapm: remove unnecessary definition ASoC: soc-dapm: remove EXPORT_SYMBOL_GPL() for snd_soc_dapm_free() ASoC: soc-dapm: remove snd_soc_dapm_nc_pin[_unlocked]() ASoC: soc-dapm: remove snd_soc_dapm_weak_routes() ASoC: soc-dapm: reordering function definitions ASoC: soc-dapm: reordering header definitions ASoC: soc-dapm: use component instead of cmpnt ASoC: soc-dapm: use common name for dapm ASoC: soc-dapm: add prefix on dapm_mark_endpoints_dirty() ASoC: soc-dapm: add prefix on dapm_xxx_event() ASoC: soc-dapm: add prefix on soc_dapm_dev_attrs ASoC: rt700: don't set dapm->bias_level ASoC: rt715: don't set dapm->bias_level ASoC: soc-dapm: set bias_level if snd_soc_dapm_set_bias_level() was successed ASoC: samsung: bell: don't set dapm->bias_level directly ASoC: samsung: speyside: don't set dapm->bias_level directly ASoC: samsung: tobermory: don't set dapm->bias_level directly ASoC: samsung: littlemill: don't set dapm->bias_level directly Laurentiu Mihalcea (1): ASoC: dt-bindings: cirrus,cs42xx8: add 'port' property Luca Weiss (6): ASoC: qcom: sm8250: set card driver name from match data ASoC: qcom: sm8250: add DisplayPort Jack support ASoC: qcom: sm8250: Add Fairphone 5 soundcard compatible ASoC: dt-bindings: qcom,q6afe: Document q6usb subnode ASoC: dt-bindings: qcom,sm8250: Add Fairphone 4 sound card ASoC: qcom: sm8250: Add Fairphone 4 soundcard compatible Mac Chiang (2): ASoC: Intel: soc-acpi-intel-ptl-match: add support ptl-rt721-l0 ASoC: Intel: soc-acpi-intel-lnl-match: add rt1320_l12_rt714_l0 support Maciej Strozek (1): ASoC: SDCA: Generic interrupt support Mark Brown (32): ASoC: codecs: wcd93xx: Few simplifications of code and ASoC: Intel: avs: Add rt5640 board support Use helper function for_each_child_of_node_scoped() ASoC: tas571x: add support for tas5753 ASoC: remove component->id Cleanup in rockchip_sai.c ASoC: Add Richtek RTQ9124 support Add DisplayPort sound support for Fairphone 5 ASoC: codecs: More const and unused member cleanups ASoC: mediatek: use reserved memory or enable ASoC: core/topology/Intel: ASoC: SOF: ipc4/Intel: Support for partial context ASoC: SOF: ipc4-pcm: Harmonize set pipeline state dbg ASoC: hdac_hdmi: Rate limit logging on connection and disconnection ASoC: codecs: wcd93xx: Drop enums for chipid values Add SoundWire machines for ACP7.0/ACP7.1 sof stack ASoC: Standardize ASoC menu ASoC: sdw_utils: generate combined spk components ASoC: Add SDCA IRQ support and some misc fixups Add sound card support for QCS8275 treewide: Remove redundant ASoC: soc-dapm: cleanups ASoC: Intel: add support for new devices ASoC: don't set dapm->bias_level directly ASoC: set bias_level at if Update SDCA Kconfig ASoC: codec: Convert to GPIO descriptors for ASoC: convert from clk round_rate() to SDCA Bug Fixes Add SDCA DAI ops helpers More minor SDCA changes ASoC: fsl_xcvr: get channel status data in two cases Pei Xiao (2): ASOC: rockchip: fix capture stream handling in rockchip_sai_xfer_stop ASOC: rockchip: Use helper function devm_clk_get_enabled() Peng Fan (4): ASoC: codec: rockchip_sai: Remove including of_gpio.h ASoC: codec: tlv320aic32x4: Drop aic32x4_pdata usage ASoC: codec: tlv320aic32x4: Sort headers alphabetically ASoC: codec: tlv320aic32x4: Convert to GPIO descriptors Peter Ujfalusi (13): ASoC: topology: Do not call snd_soc_remove_pcm_runtime() for ignored links ASoC: core: Check for rtd == NULL in snd_soc_remove_pcm_runtime() ASoC: Intel: skl_hda_dsp_generic: Implement add_dai_link to filter HDMI PCMs ASoC: Intel: sof_sdw: Implement add_dai_link to filter HDMI PCMs ASoC: SOF: ipc4-priv: Add kernel doc for fw_context_save of sof_ipc4_fw_data ASoC: SOF: ipc4/Intel: Add support for library restore firmware functionality ASoC: SOF: ipc4-pcm: Look for best matching hw_config for SSP ASoC: SOF: pcm: Remove local create_page_table() wrapper function ASoC: SOF: topology: Parse the dapm_widget_tokens in case of DSPless mode ASoC: SOF: pcm: Reverse check for prepared stream in sof_pcm_hw_params() ASoC: SOF: Intel: hda: Do not probe Soundwire in nocodec mode ASoC: SOF: ipc4-pcm: Enable delay reporting for ChainDMA streams ASoC: sdw_utils: Add missed component_name strings for speaker amps Prasad Kumpatla (2): ASoC: dt-bindings: qcom,sm8250: Add QCS8275 sound card ASoC: qcom: sc8280xp: Add support for QCS8275 Richard Fitzgerald (1): firmware: cs_dsp: Remove unused struct list_head from cs_dsp_coeff_ctl Rob Herring (Arm) (1): ASoC: Use of_reserved_mem_region_to_resource() for "memory-region" Sachin Mokashi (1): ASoC: Intel: Replace deprecated strcpy() with strscpy() Sakari Ailus (11): PM: runtime: Document return values of suspend-related API functions PM: runtime: Mark last busy stamp in pm_runtime_put_autosuspend() PM: runtime: Mark last busy stamp in pm_runtime_put_sync_autosuspend() PM: runtime: Mark last busy stamp in pm_runtime_autosuspend() PM: runtime: Mark last busy stamp in pm_request_autosuspend() Documentation: PM: *_autosuspend() functions update last busy time ASoC: atmel: Remove redundant pm_runtime_mark_last_busy() calls ASoC: codecs: Remove redundant pm_runtime_mark_last_busy() calls ASoC: Intel: Remove redundant pm_runtime_mark_last_busy() calls ASoC: component: Remove redundant pm_runtime_mark_last_busy() calls ASoC: SOF: Remove redundant pm_runtime_mark_last_busy() calls Seppo Ingalsuo (1): ASoC: SOF: ipc4-topology: Add load of ASRC component Shenghao Ding (2): ASoC: tas2781: Drop the unnecessary symbol imply ASoC: tas2781: Move the "include linux/debugfs.h" into tas2781.h Shengjiu Wang (6): ASoC: wm8524: enable constraints when sysclk is configured. ASoC: fsl_mqs: Distinguish different modules by system manager indices ASoC: fsl_mqs: rename system manager indices for i.MX95 ASoC: fsl-asoc-card: add sysclk_ratio for calculate sysclk frequency ASoC: fsl_xcvr: get channel status data when PHY is not exists ASoC: fsl_xcvr: get channel status data with firmware exists Shuming Fan (3): ASoC: SDCA: add support for HIDE entity properties and HID descriptor/report ASoC: SDCA: add a HID device for HIDE entity ASoC: SDCA: add route by the number of input pins in MU entity Srinivas Kandagatla (1): ASoC: qcom: use drvdata instead of component to keep id Stephen Rothwell (1): ASoC: SDCA: Check devm_mutex_init() return value Terry Cheong (1): ASoC: hdmi-codec: use SND_JACK_AVOUT as jack status Varshini Rajendran (1): ASoC: dt-bindings: atmel,at91-ssc: add microchip,sam9x7-ssc Venkata Prasad Potturu (4): ASoC: amd: acp: Fix pointer assignments for snd_soc_acpi_mach structures ASoC: amd: acp: Add legacy driver support acp7.2 based platforms ASoC: amd: acp: Enable I2S support for acp7.2 based platforms ASoC: amd: acp: Enable acp7.2 platform based DMIC support in machine driver Vijendar Mukunda (2): ASoC: amd: acp: add soundwire machine for ACP7.0 and ACP7.1 sof stack ASoC: SOF: amd: add alternate machines for ACP7.0 and ACP7.1 platforms Weidong Wang (2): ASoC: codecs: Add calibration function to aw88399 chip ASoC: codecs: Add acpi_match_table for aw88399 driver Xinxin Wan (1): ASoC: codecs: rt5640: Retry DEVICE_ID verification .../devicetree/bindings/sound/atmel,at91-ssc.yaml | 11 +- .../devicetree/bindings/sound/cirrus,cs42xx8.yaml | 4 + .../bindings/sound/fsl,mxs-audio-sgtl5000.yaml | 81 ++ .../bindings/sound/mediatek,mt8173-afe-pcm.yaml | 98 ++ .../devicetree/bindings/sound/mt8186-afe-pcm.yaml | 5 + .../devicetree/bindings/sound/mt8192-afe-pcm.yaml | 5 + .../devicetree/bindings/sound/mtk-afe-pcm.txt | 45 - .../bindings/sound/mxs-audio-sgtl5000.txt | 42 - .../bindings/sound/qcom,lpass-va-macro.yaml | 28 +- .../devicetree/bindings/sound/qcom,q6afe.yaml | 13 + .../devicetree/bindings/sound/qcom,sm8250.yaml | 2 + .../devicetree/bindings/sound/qcom,wcd939x.yaml | 3 + .../devicetree/bindings/sound/richtek,rt9123.yaml | 9 +- .../devicetree/bindings/sound/ti,tas57xx.yaml | 2 + Documentation/power/runtime_pm.rst | 50 +- MAINTAINERS | 11 + arch/mips/configs/generic/board-marduk.config | 1 - drivers/firmware/cirrus/cs_dsp.c | 45 +- drivers/hid/hid-core.c | 3 + include/linux/firmware/cirrus/cs_dsp.h | 2 - include/linux/firmware/imx/sm.h | 20 +- include/linux/pm_runtime.h | 187 +++- include/sound/sdca_asoc.h | 19 + include/sound/sdca_function.h | 85 +- include/sound/sdca_hid.h | 25 + include/sound/sdca_interrupts.h | 78 ++ include/sound/soc-component.h | 1 - include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 236 +++-- include/sound/soc_sdw_utils.h | 1 + include/sound/sof/ipc4/header.h | 2 + include/sound/tas2781.h | 4 + include/sound/tlv320aic32x4.h | 9 - include/uapi/linux/input.h | 1 + sound/soc/Kconfig | 8 +- sound/soc/adi/Kconfig | 9 +- sound/soc/amd/Kconfig | 3 + sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-i2s.c | 3 + sound/soc/amd/acp/acp-legacy-common.c | 2 + sound/soc/amd/acp/acp-mach-common.c | 1 + sound/soc/amd/acp/acp-pci.c | 9 +- sound/soc/amd/acp/acp-platform.c | 2 + sound/soc/amd/acp/acp70.c | 1 + sound/soc/amd/acp/acp_common.h | 1 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 12 + sound/soc/amd/acp/amd-acpi-mach.c | 4 +- sound/soc/amd/acp/amd.h | 8 +- sound/soc/amd/mach-config.h | 1 + sound/soc/apple/Kconfig | 4 + sound/soc/atmel/Kconfig | 11 +- sound/soc/atmel/mchp-spdifrx.c | 6 - sound/soc/au1x/Kconfig | 4 + sound/soc/bcm/Kconfig | 4 + sound/soc/cirrus/Kconfig | 3 + sound/soc/codecs/Kconfig | 12 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/arizona-jack.c | 2 - sound/soc/codecs/aw88395/aw88395_device.h | 21 + sound/soc/codecs/aw88399.c | 449 ++++++++- sound/soc/codecs/aw88399.h | 34 + sound/soc/codecs/cros_ec_codec.c | 30 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41.c | 2 - sound/soc/codecs/cs35l45.c | 1 - sound/soc/codecs/cs35l56-sdw.c | 1 - sound/soc/codecs/cs35l56.c | 1 - sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l43-jack.c | 5 - sound/soc/codecs/cs42l43.c | 2 - sound/soc/codecs/cs48l32.c | 1 - sound/soc/codecs/da7218.c | 2 +- sound/soc/codecs/da7219.c | 68 +- sound/soc/codecs/es8375.c | 2 +- sound/soc/codecs/hda.c | 3 - sound/soc/codecs/hdac_hdmi.c | 307 +----- sound/soc/codecs/hdac_hdmi.h | 10 - sound/soc/codecs/hdmi-codec.c | 6 +- sound/soc/codecs/lpass-rx-macro.c | 1 - sound/soc/codecs/lpass-tx-macro.c | 1 - sound/soc/codecs/lpass-va-macro.c | 1 - sound/soc/codecs/lpass-wsa-macro.c | 1 - sound/soc/codecs/max98363.c | 3 +- sound/soc/codecs/max98373-i2c.c | 2 +- sound/soc/codecs/max98373-sdw.c | 3 +- sound/soc/codecs/max98388.c | 2 +- sound/soc/codecs/max98390.c | 2 +- sound/soc/codecs/max98396.c | 4 +- sound/soc/codecs/max98504.c | 2 +- sound/soc/codecs/max98520.c | 2 +- sound/soc/codecs/max98927.c | 2 +- sound/soc/codecs/rt1017-sdca-sdw.c | 1 - sound/soc/codecs/rt1308-sdw.c | 1 - sound/soc/codecs/rt1316-sdw.c | 1 - sound/soc/codecs/rt1318-sdw.c | 1 - sound/soc/codecs/rt1320-sdw.c | 1 - sound/soc/codecs/rt5640.c | 43 +- sound/soc/codecs/rt5640.h | 6 +- sound/soc/codecs/rt5682-sdw.c | 1 - sound/soc/codecs/rt5682.c | 26 +- sound/soc/codecs/rt5682s.c | 26 +- sound/soc/codecs/rt700.c | 4 +- sound/soc/codecs/rt711-sdca.c | 2 - sound/soc/codecs/rt711.c | 2 - sound/soc/codecs/rt712-sdca-dmic.c | 1 - sound/soc/codecs/rt712-sdca.c | 2 - sound/soc/codecs/rt715-sdca.c | 1 - sound/soc/codecs/rt715.c | 3 +- sound/soc/codecs/rt721-sdca.c | 2 - sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca.c | 2 - sound/soc/codecs/rt9123.c | 3 - sound/soc/codecs/rtq9124.c | 543 +++++++++++ sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2781-i2c.c | 3 - sound/soc/codecs/tas571x.c | 52 + sound/soc/codecs/tlv320aic32x4.c | 53 +- sound/soc/codecs/wcd-mbhc-v2.c | 2 - sound/soc/codecs/wcd9335.c | 5 - sound/soc/codecs/wcd934x.c | 4 - sound/soc/codecs/wcd937x.c | 38 +- sound/soc/codecs/wcd937x.h | 1 - sound/soc/codecs/wcd938x.c | 64 +- sound/soc/codecs/wcd939x.c | 69 +- sound/soc/codecs/wm8524.c | 55 +- sound/soc/codecs/wsa881x.c | 5 +- sound/soc/codecs/wsa883x.c | 3 +- sound/soc/codecs/wsa884x.c | 3 +- sound/soc/dwc/Kconfig | 3 + sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl-asoc-card.c | 13 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_mqs.c | 11 +- sound/soc/fsl/fsl_sai.c | 4 +- sound/soc/fsl/fsl_xcvr.c | 25 +- sound/soc/fsl/imx-card.c | 53 +- sound/soc/generic/Kconfig | 4 + sound/soc/google/Kconfig | 3 + sound/soc/hisilicon/Kconfig | 4 + sound/soc/img/Kconfig | 14 +- sound/soc/intel/Kconfig | 4 + sound/soc/intel/atom/sst/sst_pvt.c | 1 - sound/soc/intel/avs/board_selection.c | 27 + sound/soc/intel/avs/boards/Kconfig | 12 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt5640.c | 270 ++++++ sound/soc/intel/avs/core.c | 1 - sound/soc/intel/avs/debugfs.c | 2 - sound/soc/intel/avs/ipc.c | 1 - sound/soc/intel/avs/pcm.c | 1 - sound/soc/intel/boards/cht_bsw_rt5672.c | 3 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 13 + sound/soc/intel/boards/sof_es8336.c | 10 + sound/soc/intel/boards/sof_rt5682.c | 7 + sound/soc/intel/boards/sof_sdw.c | 14 + sound/soc/intel/catpt/pcm.c | 6 - sound/soc/intel/catpt/sysfs.c | 1 - sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 34 + sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 57 ++ sound/soc/jz4740/Kconfig | 4 + sound/soc/kirkwood/Kconfig | 3 + sound/soc/loongson/Kconfig | 2 +- sound/soc/mediatek/Kconfig | 4 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 4 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 66 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 56 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 7 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 4 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 7 + sound/soc/meson/Kconfig | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 +- sound/soc/mxs/Kconfig | 2 +- sound/soc/pxa/Kconfig | 4 + sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/lpass-platform.c | 27 +- sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 8 +- sound/soc/qcom/sc8280xp.c | 1 + sound/soc/qcom/sm8250.c | 28 +- sound/soc/renesas/Kconfig | 2 +- sound/soc/renesas/rcar/core.c | 35 +- sound/soc/renesas/rcar/ctu.c | 8 +- sound/soc/renesas/rcar/dma.c | 4 +- sound/soc/renesas/rcar/dvc.c | 8 +- sound/soc/renesas/rcar/mix.c | 8 +- sound/soc/renesas/rcar/msiof.c | 94 +- sound/soc/renesas/rcar/src.c | 10 +- sound/soc/renesas/rcar/ssi.c | 18 +- sound/soc/renesas/rcar/ssiu.c | 7 +- sound/soc/rockchip/Kconfig | 23 +- sound/soc/rockchip/rockchip_sai.c | 52 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/bells.c | 2 - sound/soc/samsung/littlemill.c | 2 - sound/soc/samsung/speyside.c | 2 - sound/soc/samsung/tobermory.c | 2 - sound/soc/sdca/Kconfig | 20 + sound/soc/sdca/Makefile | 6 +- sound/soc/sdca/sdca_asoc.c | 469 +++++++-- sound/soc/sdca/sdca_functions.c | 264 ++++- sound/soc/sdca/sdca_hid.c | 127 +++ sound/soc/sdca/sdca_interrupts.c | 444 +++++++++ sound/soc/sdca/sdca_regmap.c | 29 +- sound/soc/sdw_utils/soc_sdw_cs_amp.c | 10 - sound/soc/sdw_utils/soc_sdw_maxim.c | 9 - sound/soc/sdw_utils/soc_sdw_rt_amp.c | 6 - sound/soc/sdw_utils/soc_sdw_rt_mf_sdca.c | 7 - sound/soc/sdw_utils/soc_sdw_utils.c | 38 + sound/soc/soc-component.c | 1 - sound/soc/soc-core.c | 25 +- sound/soc/soc-dai.c | 16 +- sound/soc/soc-dapm.c | 1018 +++++++++----------- sound/soc/soc-ops-test.c | 29 +- sound/soc/soc-ops.c | 26 +- sound/soc/soc-topology.c | 7 +- sound/soc/sof/Kconfig | 2 +- sound/soc/sof/amd/pci-acp70.c | 1 + sound/soc/sof/control.c | 1 - sound/soc/sof/debug.c | 1 - sound/soc/sof/imx/imx-common.c | 36 +- sound/soc/sof/intel/hda-loader.c | 7 +- sound/soc/sof/intel/hda.c | 5 + sound/soc/sof/ipc3-dtrace.c | 1 - sound/soc/sof/ipc4-loader.c | 7 +- sound/soc/sof/ipc4-pcm.c | 137 ++- sound/soc/sof/ipc4-priv.h | 6 + sound/soc/sof/ipc4-topology.c | 94 +- sound/soc/sof/ipc4-topology.h | 25 + sound/soc/sof/ipc4.c | 39 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 11 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 11 +- sound/soc/sof/pcm.c | 27 +- sound/soc/sof/sof-client-ipc-flood-test.c | 1 - sound/soc/sof/sof-client-ipc-kernel-injector.c | 1 - sound/soc/sof/sof-client-ipc-msg-injector.c | 1 - sound/soc/sof/sof-client-probes.c | 3 - sound/soc/sof/topology.c | 15 +- sound/soc/spear/Kconfig | 4 + sound/soc/sprd/Kconfig | 4 + sound/soc/starfive/Kconfig | 12 +- sound/soc/sti/Kconfig | 2 +- sound/soc/stm/Kconfig | 2 +- sound/soc/stm/stm32_i2s.c | 21 +- sound/soc/stm/stm32_sai_sub.c | 14 +- sound/soc/sunxi/Kconfig | 2 +- sound/soc/tegra/Kconfig | 4 + sound/soc/tegra/tegra210_ahub.c | 2 +- sound/soc/ti/Kconfig | 2 +- sound/soc/uniphier/Kconfig | 14 +- sound/soc/ux500/Kconfig | 2 +- sound/soc/xilinx/Kconfig | 4 + sound/soc/xtensa/Kconfig | 4 + 253 files changed, 5171 insertions(+), 2291 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,mxs-audio-sgtl5000.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8173-afe-pcm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt delete mode 100644 Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt create mode 100644 include/sound/sdca_hid.h create mode 100644 include/sound/sdca_interrupts.h delete mode 100644 sound/soc/codecs/hdac_hdmi.h create mode 100644 sound/soc/codecs/rtq9124.c create mode 100644 sound/soc/intel/avs/boards/rt5640.c create mode 100644 sound/soc/sdca/sdca_hid.c create mode 100644 sound/soc/sdca/sdca_interrupts.c

2 1

Re: [PATCH v3 2/4] dt-bindings: sound: bindings for tac5x1x family of codecs
by Holalu Yogendra, Niranjan 28 Jul '25

28 Jul '25

Hi Krzysztof, Thanks for the detailed review comments. Request you to take a look at comments below. On Fri, Jun 27, 2025 at 1:18 PM +0530, Krzysztof Kozlowski wrote: > On Thu, Jun 26, 2025 at 11:43:31PM +0530, Niranjan H Y wrote: > > + ti,vref: > > + description: VREF required voltage. If node is omitted then VREF is > Voltage is in microvolt. Use proper suffix/values. Is this convention ? Is there any document on these I can refer ? > > > + oneOf: > > + - const: 0 > > + description: VREF output is powered to 2.75V. > > + - const: 1 > > + description: VREF output is powered to 2.5V. > > + - const: 2 > > + description: VREF output is powered to 1.375V. > > + ti,gpios-func: > > + description: | > > + Array indicating the GPIO1, GPIO2, GPO1 Functionality in the same order. > > + Each integer elemnent in the array represent the following > > + - 0 TAC5X1X_GPIO_DISABLE - GPIO is Disabled > > + - 1 TAC5X1X_GPIO_GPI - General Purpose Input > > + - 2 ADC3XXX_GPIO_GPO - General Purpose Output > > + - 3 TAC5X1X_GPIO_IRQ - Chip Interrupt > > + - 4 TAC5X1X_GPIO_PDMCLK - PDM CLK Output > > + - 5 TAC5X1X_GPIO_P_DOUT - Primary ASI DOUT > > + - 6 TAC5X1X_GPIO_P_DOUT2 - Primary ASI DOUT2 > > + - 7 TAC5X1X_GPIO_S_DOUT - Secondary ASI DOUT > > + - 8 TAC5X1X_GPIO_S_DOUT2 - Secondary ASI DOUT2 > > + - 9 TAC5X1X_GPIO_S_BCLK - Secondary BCLK Output > > + - 10 TAC5X1X_GPIO_S_FSYNC - Secondary FSYNC Output > > + - 11 TAC5X1X_GPIO_CLKOUT - General Purpose Output > > + - 12 TAC5X1X_GPIO_DOUT_MUX > > + - 13 TAC5X1X_GPIO_DAISY_OUT > > + $ref: /schemas/types.yaml#/definitions/uint32-array > > + minItems: 3 > > + maxItems: 3 > > Don't encode gpios and pins as some custom properties. Could you please suggest what is the alternative. Please suggest some example which I can refer ? Regards Niranjan

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[PATCH v2] ALSA: hda: Add TAS2770 support
by Baojun Xu 26 Jul '25

26 Jul '25

Add TAS2770 support in TI's HDA driver. And add hda_chip_id for more products. Distinguish DSP and non-DSP in firmware loading function. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2: - Rewrite the patch description - Move registers define for TAS2770 from tas2781.h to tas2770-tlv.h - Add new chip ID define for HDA - Add digital volume control in tas2770_snd_controls - Change public kcontrol name from tas27xx_ to tasdevice_ - Create new function for kcontrol adding - Create new function for DSP binary loading - Rename chip_id to hda_chip_id in i2c_probe for the supported devices --- include/sound/tas2770-tlv.h | 23 +++ .../hda/codecs/side-codecs/tas2781_hda_i2c.c | 161 +++++++++++------- 2 files changed, 127 insertions(+), 57 deletions(-) create mode 100644 include/sound/tas2770-tlv.h diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h new file mode 100644 index 000000000000..c0bd495b4a07 --- /dev/null +++ b/include/sound/tas2770-tlv.h @@ -0,0 +1,23 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2770 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2770 hda driver implements for one, two, or even multiple +// TAS2770 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2770_TLV_H__ +#define __TAS2770_TLV_H__ + +#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); + +#endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index bacc3f6ed4bd..65a48326f5b9 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -24,6 +24,7 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2770-tlv.h> #include <sound/tas2781-tlv.h> #include "hda_local.h" @@ -45,9 +46,18 @@ #define TAS2563_CAL_TLIM TASDEVICE_REG(0, 0x10, 0x14) #define TAS2563_CAL_R0 TASDEVICE_REG(0, 0x0f, 0x34) +enum device_chip_id { + HDA_TAS2563, + HDA_TAS2770, + HDA_TAS2781, + HDA_OTHERS +}; + struct tas2781_hda_i2c_priv { struct snd_kcontrol *snd_ctls[2]; int (*save_calibration)(struct tas2781_hda *h); + + int hda_chip_id; }; static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) @@ -245,6 +255,15 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static const struct snd_kcontrol_new tas2770_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2770_AMP_LEVEL, + 0, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, tas2770_amp_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2770_DVC_LEVEL, + 0, 0, 31, 0, tas2781_amp_getvol, + tas2781_amp_putvol, tas2770_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -253,7 +272,7 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_force_fwload_get, tas2781_force_fwload_put), }; -static const struct snd_kcontrol_new tas2781_prof_ctrl = { +static const struct snd_kcontrol_new tasdevice_prof_ctrl = { .name = "Speaker Profile Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_profile, @@ -261,7 +280,7 @@ static const struct snd_kcontrol_new tas2781_prof_ctrl = { .put = tasdevice_set_profile_id, }; -static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { +static const struct snd_kcontrol_new tasdevice_dsp_prog_ctrl = { .name = "Speaker Program Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_programs, @@ -269,7 +288,7 @@ static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { .put = tasdevice_program_put, }; -static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl = { +static const struct snd_kcontrol_new tasdevice_dsp_conf_ctrl = { .name = "Speaker Config Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_config, @@ -378,44 +397,34 @@ static void tas2781_hda_remove_controls(struct tas2781_hda *tas_hda) snd_ctl_remove(codec->card, tas_hda->prof_ctl); } -static void tasdev_fw_ready(const struct firmware *fmw, void *context) +static void tasdev_add_kcontrols(struct tasdevice_priv *tas_priv, + struct snd_kcontrol **ctls, struct hda_codec *codec, + const struct snd_kcontrol_new *tas_snd_ctrls, int num_ctls) { - struct tasdevice_priv *tas_priv = context; - struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); - struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; - struct hda_codec *codec = tas_priv->codec; - int i, ret, spk_id; - - pm_runtime_get_sync(tas_priv->dev); - mutex_lock(&tas_priv->codec_lock); + int i, ret; - ret = tasdevice_rca_parser(tas_priv, fmw); - if (ret) - goto out; - - tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl, tas_priv); - ret = snd_ctl_add(codec->card, tas_hda->prof_ctl); - if (ret) { - dev_err(tas_priv->dev, - "Failed to add KControl %s = %d\n", - tas2781_prof_ctrl.name, ret); - goto out; - } - - for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { - hda_priv->snd_ctls[i] = snd_ctl_new1(&tas2781_snd_controls[i], - tas_priv); - ret = snd_ctl_add(codec->card, hda_priv->snd_ctls[i]); + for (i = 0; i < num_ctls; i++) { + ctls[i] = snd_ctl_new1( + &tas_snd_ctrls[i], tas_priv); + ret = snd_ctl_add(codec->card, ctls[i]); if (ret) { dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", - tas2781_snd_controls[i].name, ret); - goto out; + tas_snd_ctrls[i].name, ret); + break; } } +} - tasdevice_dsp_remove(tas_priv); +static void tasdevice_dspfw_init(void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); + struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; + struct hda_codec *codec = tas_priv->codec; + int ret, spk_id; + tasdevice_dsp_remove(tas_priv); tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; if (tas_priv->speaker_id != NULL) { // Speaker id need to be checked for ASUS only. @@ -441,28 +450,12 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - goto out; - } - - tas_hda->dsp_prog_ctl = snd_ctl_new1(&tas2781_dsp_prog_ctrl, - tas_priv); - ret = snd_ctl_add(codec->card, tas_hda->dsp_prog_ctl); - if (ret) { - dev_err(tas_priv->dev, - "Failed to add KControl %s = %d\n", - tas2781_dsp_prog_ctrl.name, ret); - goto out; - } - - tas_hda->dsp_conf_ctl = snd_ctl_new1(&tas2781_dsp_conf_ctrl, - tas_priv); - ret = snd_ctl_add(codec->card, tas_hda->dsp_conf_ctl); - if (ret) { - dev_err(tas_priv->dev, - "Failed to add KControl %s = %d\n", - tas2781_dsp_conf_ctrl.name, ret); - goto out; + return; } + tasdev_add_kcontrols(tas_priv, &tas_hda->dsp_prog_ctl, codec, + &tasdevice_dsp_prog_ctrl, 1); + tasdev_add_kcontrols(tas_priv, &tas_hda->dsp_conf_ctl, codec, + &tasdevice_dsp_conf_ctrl, 1); tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; tasdevice_prmg_load(tas_priv, 0); @@ -475,9 +468,45 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) * calibrated data inside algo. */ hda_priv->save_calibration(tas_hda); +} + +static void tasdev_fw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); + struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; + struct hda_codec *codec = tas_priv->codec; + int ret; + + pm_runtime_get_sync(tas_priv->dev); + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; - tasdevice_tuning_switch(tas_hda->priv, 0); - tas_hda->priv->playback_started = true; + tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + tasdev_add_kcontrols(tas_priv, &tas_hda->prof_ctl, codec, + &tasdevice_prof_ctrl, 1); + + switch (hda_priv->hda_chip_id) { + case HDA_TAS2770: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas2770_snd_controls[0], + ARRAY_SIZE(tas2770_snd_controls)); + break; + case HDA_TAS2781: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas2781_snd_controls[0], + ARRAY_SIZE(tas2781_snd_controls)); + tasdevice_dspfw_init(context); + break; + case HDA_TAS2563: + tasdevice_dspfw_init(context); + break; + default: + break; + } out: mutex_unlock(&tas_hda->priv->codec_lock); @@ -581,16 +610,33 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) return -ENOMEM; if (strstr(dev_name(&clt->dev), "TIAS2781")) { - device_name = "TIAS2781"; + /* + * TAS2781, integrated on-chip DSP with + * global I2C address supported. + */ + device_name = "TIAS2781"; + hda_priv->hda_chip_id = HDA_TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; + } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2770")) { + /* + * TAS2770, has no on-chip DSP, so no calibration data + * required; has no global I2C address supported. + */ + device_name = "TXNW2770"; + hda_priv->hda_chip_id = HDA_TAS2770; } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2781:00-tas2781-hda.0")) { device_name = "TXNW2781"; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else if (strstr(dev_name(&clt->dev), "INT8866")) { - device_name = "INT8866"; + /* + * TAS2563, integrated on-chip DSP with + * global I2C address supported. + */ + device_name = "INT8866"; + hda_priv->hda_chip_id = HDA_TAS2563; hda_priv->save_calibration = tas2563_save_calibration; tas_hda->priv->global_addr = TAS2563_GLOBAL_ADDR; } else { @@ -727,6 +773,7 @@ static const struct i2c_device_id tas2781_hda_i2c_id[] = { static const struct acpi_device_id tas2781_acpi_hda_match[] = { {"INT8866", 0 }, {"TIAS2781", 0 }, + {"TXNW2770", 0 }, {"TXNW2781", 0 }, {} }; -- 2.43.0

3 2

tinycompress: add support for new SNDRV_COMPRESS_TSTAMP64 and _AVAIL64
by GitHub pull_request - opened 25 Jul '25

25 Jul '25

alsa-project/tinycompress pull request #29 was opened from jorisverhaegen: These 2 ioctl are under review to be added to kernel. See https://lore.kernel.org/linux-sound/20250725114249.2086974-1-verhaegen@goog… This pull request should be merged after this kernel is available. Request URL : https://github.com/alsa-project/tinycompress/pull/29 Patch URL : https://github.com/alsa-project/tinycompress/pull/29.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0