[lgirdwood]

Summary

Follow-up improvements for the WebRTC audio processing module suite merged in #11044:

1. audio: webrtc_agc: add 32 kHz and 48 kHz sample rate support

   • WebRtcAgc_ProcessDigital() natively accepts 8, 16, 32, and 48 kHz sample rates using 160-sample sub-bands (num_bands = 1 at 8/16 kHz, 2 at 32 kHz, 3 at 48 kHz).
   • Extend webrtc_agc and webrtc-agc.conf to support 32 kHz and 48 kHz streams alongside 8 kHz and 16 kHz by downsampling each 10 ms frame to 16 kHz for VAD/envelope tracking, running a separate full-rate overload limiter pass across all 10 subframes, and applying piecewise-linear Q16 gain interpolation directly to the full-rate samples.

2. audio: webrtc_aec: configure module for DP domain scheduling

   • Configure the webrtc_aec LLEXT manifest (.domain_dp = 1), rimage TOML (domain_types = "1"), and topology2 widget definition (scheduler_domain "DP" with stack_bytes_requirement 16384) to run in the Data Processing (DP) scheduling domain rather than on the 1 ms Low Latency (LL) thread.
   • Add webrtc_aec_is_ready_to_process() so the DP scheduler triggers processing whenever a full 10 ms microphone capture frame is available and output space is free, without blocking when the playback echo reference stream is idle.

3. app: add WebRTC Kconfig overlays and COMP_STUBS defaults

   • Default webrtc_aec, webrtc_agc, and webrtc_hpf to their pass-through stub backends when CONFIG_COMP_STUBS=y (matching webrtc_ns, webrtc_ns2, and webrtc_vad), and enable all six WebRTC modules in app/stub_build_all_ipc4.conf.
   • Add app/webrtc_overlay.conf (LLEXT real backends + heap sizing) and app/webrtc_stubs_overlay.conf (LLEXT stub backends).

[lgirdwood]

Commit messages are not line wrapped correctly, and webrtc modules will be deployed as llext raher than built in so we need to make sure we can build as llext.

[lgirdwood]

Updated the branch to address the review comments:

1. Wrapped all commit messages to <= 72 characters per line.
2. Added detailed step-by-step inline comments (2a–2e) to the 32/48 kHz multi-band block in src/audio/webrtc_agc/webrtc_agc-agc.c.
3. Configured all six WebRTC modules (webrtc_vad, webrtc_ns, webrtc_aec, webrtc_ns2, webrtc_agc, webrtc_hpf) as LLEXT modules (=m) across app/stub_build_all_ipc4.conf, app/webrtc_overlay.conf, and app/webrtc_stubs_overlay.conf, and verified PTL LLEXT builds (sof-ptl-openmodules.ri) in both real-backend and stub modes.

[Copilot]

🟡 Changes recommended

Critical issues can silently discard 240 audio samples during consecutive 48 kHz calls and leave the advertised real-AEC overlay using a pass-through stub, while unfiltered band splitting also distorts VAD-driven gain decisions.

3 open issues

• Overlay selects the AEC stub instead of a real backend · New
• 15 ms calls overflow the 48 kHz input FIFO · New
• Unfiltered band splitting corrupts VAD input at high sample rates · New

What changed in this PR

Extends SOF’s WebRTC AGC to higher sample rates and adds configuration options for WebRTC deployment and scheduler stack sizing.

Changes:

• Adds 32/48 kHz AGC processing and matching topology formats.
• Makes the per-core LL scheduler stack configurable, with a larger default for AEC3.
• Adds real and stub WebRTC overlays and enables all six modules in the IPC4 stub configuration.

File	Description
tools/​topology/​topology2/​include/​components/​webrtc-agc.conf	Adds 32/48 kHz input and output formats.
src/​schedule/​zephyr_domain.c	Uses the configurable LL stack size.
src/​schedule/​Kconfig	Defines LL stack sizing and the AEC3 default.
src/​audio/​webrtc_hpf/​Kconfig	Defaults HPF to its stub under COMP_STUBS.
src/​audio/​webrtc_agc/​webrtc_agc.c	Accepts 32/48 kHz streams.
src/​audio/​webrtc_agc/​webrtc_agc-agc.c	Adds phase splitting and peak ordering for higher rates.
src/​audio/​webrtc_agc/​README.md	Documents expanded frame sizes and rates.
src/​audio/​webrtc_agc/​Kconfig	Defaults AGC to its stub under COMP_STUBS.
src/​audio/​webrtc_aec/​Kconfig	Defaults AEC to its stub under COMP_STUBS.
app/​webrtc_stubs_overlay.conf	Adds a six-module pass-through overlay.
app/​webrtc_overlay.conf	Adds real-backend settings and heap sizing.
app/​stub_build_all_ipc4.conf	Enables all six WebRTC modules.

_{Review effort: Balanced}

[Copilot]

🟡 Changes recommended

Two critical AGC issues can overflow audio and probe calculations, causing wrapped output samples or reconstructed negative gains.

2 open issues

• Bound probe amplitude by the maximum gain-table entry · New
• Use 64-bit multiplication before saturating scaled samples · New

3 resolved since last review

• 15 ms calls overflow the 48 kHz input FIFO
• Overlay selects the AEC stub instead of a real backend
• Unfiltered band splitting corrupts VAD input at high sample rates

_{Review effort: Balanced}

[Copilot]

Note

Copilot was unable to run its full agentic suite in this review.

Copilot review overview

5 open issues

• webrtc_agc_process_multiband() always calls WebRtcAgc_ProcessDigital() with num_bands = 2… · New
• The loop consumes input (offset += chunk) even when there isn’t enough processed output available… · New
• Use 64-bit multiplication before saturating scaled samples
• Bound probe amplitude by the maximum gain-table entry
• This option is explicitly Zephyr-specific (and is consumed in zephyr_domain.c), but it’s… · New

_{Review effort: Lite}

[intel-sofci]

PR 11419: test results

Run date: 2026-11-09 10:58 UTC

Tested commit: 1bf9c2043f1ab8f06687a154e8bd3146337b6a0c

[lgirdwood]

Unrelated CI failures. Feature not yet tested in CI.