feat(linux): WebRTC AEC3 capture/playback backend with render-side tee
Adds gold-standard Linux echo cancellation: in-app WebRTC AEC3 (Audio Processing Module) via the webrtc-audio-processing crate, using the same algorithm as Chrome WebRTC, Zoom, Teams, and Jitsi. Runs entirely in-process, so it works identically on ALSA / PulseAudio / PipeWire systems — no dependency on user-configured echo-cancel modules. Architecture: - New crates/wzp-client/src/audio_linux_aec.rs module (~470 lines). Contains LinuxAecCapture and LinuxAecPlayback, both using CPAL under the hood but routing samples through a shared Arc<webrtc_audio_processing::Processor>. The playback path tees each 20 ms frame into APM.process_render_frame as the echo reference BEFORE handing the samples to CPAL's output callback. The capture path runs APM.process_capture_frame on each mic frame in place before pushing to the audio ring buffer. This is the "tee the playback ring" approach that Zoom/Teams/Jitsi use. - New `linux-aec` feature in wzp-client pulling in the webrtc-audio-processing crate at v2.x with the `bundled` sub-feature. Bundled means the vendored PulseAudio WebRTC C++ sources are statically compiled via meson+ninja at cargo build time — no runtime .so dependency, avoids Debian Bookworm's stale libwebrtc-audio-processing-dev 0.3 package (which predates AEC3). Dep is target-gated to Linux, so enabling the feature on non-Linux is a no-op. - lib.rs re-exports LinuxAecCapture/LinuxAecPlayback as AudioCapture/AudioPlayback when `linux-aec` is on, otherwise falls back to the CPAL audio_io path. Shared public API (start/ring/stop/Drop) means downstream code is unchanged. - New `linux-aec` feature in wzp-desktop forwards to wzp-client/linux-aec so `cargo tauri build -- --features wzp-desktop/linux-aec` builds the AEC variant. APM configuration: - EchoCancellation: High suppression, delay-agnostic mode on, extended filter on, stream_delay_ms=60 initial hint - NoiseSuppression: High - HighPassFilter: on - AGC: off (can fight Opus encoder's own gain staging + adaptive quality controller; add later if users report low mic level) Frame size handling: - Pipeline uses 20 ms frames (960 samples @ 48 kHz mono) - APM requires strict 10 ms (480 samples) per call - Each 20 ms frame is split into two 480-sample halves, APM called twice, halves stitched back - Same pattern for render and capture sides - Carry-buffer logic handles the case where CPAL delivers samples in arbitrary chunk sizes that don't divide 960 Build infrastructure: - scripts/Dockerfile.linux-desktop-builder adds meson, ninja-build, python3, clang for the webrtc-audio-processing bundled build - scripts/build-linux-desktop-docker.sh takes a new --aec flag that enables the linux-aec feature and renames the output artifacts with an `-aec` suffix so noAEC and AEC variants can coexist on disk Task #30. Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
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@@ -21,6 +21,11 @@ pub mod audio_vpio;
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// this feature on non-Windows targets is a no-op.
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#[cfg(all(feature = "windows-aec", target_os = "windows"))]
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pub mod audio_wasapi;
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// WebRTC AEC3 (Audio Processing Module) wrapper around CPAL capture + playback
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// on Linux. Only compiled when `linux-aec` feature is on AND target is Linux.
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// The webrtc-audio-processing dep is itself gated to Linux in Cargo.toml.
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#[cfg(all(feature = "linux-aec", target_os = "linux"))]
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pub mod audio_linux_aec;
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pub mod bench;
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pub mod call;
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pub mod drift_test;
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@@ -30,24 +35,48 @@ pub mod handshake;
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pub mod metrics;
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pub mod sweep;
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// AudioPlayback always comes from the CPAL path (`audio_io`). We do not
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// need OS-level processing on the playback side because Windows's
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// communications AEC, once engaged on the capture stream, uses the system
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// render mix as the reference signal — it cancels echo from CPAL playback
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// (and any other app's audio) without special handling.
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#[cfg(feature = "audio")]
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pub use audio_io::AudioPlayback;
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// AudioPlayback: three possible backends depending on feature flags.
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// 1. Default CPAL (`audio_io::AudioPlayback`) — baseline on every platform.
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// 2. Linux AEC (`audio_linux_aec::LinuxAecPlayback`) — CPAL + WebRTC APM
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// render-side tee, so echo from speakers gets cancelled from the mic.
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//
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// On macOS and Windows we always use the default CPAL playback because:
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// - macOS: VoiceProcessingIO handles AEC at the capture side (Apple's
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// native hardware AEC uses its own reference signal handling).
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// - Windows: WASAPI AudioCategory_Communications AEC uses the system
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// render mix as reference — no per-process plumbing needed.
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//
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// Linux is the only platform where the in-app approach is necessary, so
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// the AEC playback path is gated to target_os = "linux".
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// AudioCapture: two possible backends. Windows-AEC path when compiled in,
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// otherwise the plain CPAL path. The two types share the same public API
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// (`start`, `ring`, `stop`, `Drop`) so downstream code is identical.
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#[cfg(all(
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feature = "audio",
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any(not(feature = "windows-aec"), not(target_os = "windows"))
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any(not(feature = "linux-aec"), not(target_os = "linux"))
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))]
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pub use audio_io::AudioPlayback;
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#[cfg(all(feature = "linux-aec", target_os = "linux"))]
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pub use audio_linux_aec::LinuxAecPlayback as AudioPlayback;
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// AudioCapture: three possible backends depending on feature flags.
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// 1. Default CPAL (`audio_io::AudioCapture`) — baseline on every platform.
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// 2. Windows AEC (`audio_wasapi::WasapiAudioCapture`) — direct WASAPI
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// with AudioCategory_Communications, OS APO chain does AEC.
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// 3. Linux AEC (`audio_linux_aec::LinuxAecCapture`) — CPAL + WebRTC APM
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// capture-side echo cancellation using the playback tee as reference.
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// All three expose the same public API (`start`, `ring`, `stop`, `Drop`).
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#[cfg(all(
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feature = "audio",
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any(not(feature = "windows-aec"), not(target_os = "windows")),
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any(not(feature = "linux-aec"), not(target_os = "linux"))
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))]
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pub use audio_io::AudioCapture;
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#[cfg(all(feature = "windows-aec", target_os = "windows"))]
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pub use audio_wasapi::WasapiAudioCapture as AudioCapture;
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#[cfg(all(feature = "linux-aec", target_os = "linux"))]
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pub use audio_linux_aec::LinuxAecCapture as AudioCapture;
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pub use call::{CallConfig, CallDecoder, CallEncoder};
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pub use handshake::perform_handshake;
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