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>

crates/wzp-client/src/audio_linux_aec.rs

@@ -0,0 +1,512 @@
+//! Linux AEC backend: CPAL capture + playback wired through the WebRTC Audio
+//! Processing Module (AEC3 + noise suppression + high-pass filter).
+//!
+//! This is the same algorithm used by Chrome WebRTC, Zoom, Teams, Jitsi, and
+//! any other "serious" Linux VoIP app. It runs in-process — no dependency on
+//! PulseAudio's module-echo-cancel or PipeWire's filter-chain, so it works
+//! identically on ALSA / PulseAudio / PipeWire systems.
+//!
+//! ## Architecture
+//!
+//! A single module-level `Arc<Processor>` is shared between the capture and
+//! playback paths. On each 20 ms frame (960 samples @ 48 kHz mono):
+//!
+//! - **Playback path**: `LinuxAecPlayback::start` spawns the usual CPAL
+//!   output thread, but wraps each chunk in a call to
+//!   `Processor::process_render_frame` **before** handing it to CPAL. That
+//!   gives APM an authoritative reference of exactly what's going out to
+//!   the speakers (same approach Zoom/Teams/Jitsi use). The AEC then knows
+//!   what to cancel when it sees echo in the capture stream.
+//!
+//! - **Capture path**: `LinuxAecCapture::start` spawns the usual CPAL
+//!   input thread, and runs `Processor::process_capture_frame` on each
+//!   incoming mic chunk **in place** before pushing it into the ring
+//!   buffer. The AEC subtracts the echo using the render reference it
+//!   saw on the playback side.
+//!
+//! APM is strict about frame size: it requires exactly 10 ms = 480 samples
+//! per call at 48 kHz. Our pipeline uses 20 ms = 960 samples, so each 20 ms
+//! frame is split into two 480-sample halves, APM is called twice, and the
+//! halves are stitched back together.
+//!
+//! APM only accepts f32 samples in `[-1.0, 1.0]`, so we convert i16 → f32
+//! before the call and f32 → i16 after (with clamping on the return path).
+//!
+//! ## Stream delay
+//!
+//! AEC needs to know roughly how long it takes between a sample being passed
+//! to `process_render_frame` and its echo showing up at `process_capture_frame`
+//! — i.e. the round trip through CPAL playback → speaker → air → microphone
+//! → CPAL capture. AEC3's internal estimator tracks this within a window
+//! around whatever hint we give it. We hardcode 60 ms as a reasonable
+//! starting point for typical Linux audio stacks; the delay estimator does
+//! the fine-tuning automatically.
+//!
+//! ## Thread safety
+//!
+//! `webrtc_audio_processing::Processor` is `Send + Sync` with `&self`
+//! methods. Capture and playback threads both hold an `Arc<Processor>` and
+//! call APM concurrently — the underlying C++ code serializes internally.
+
+use std::sync::atomic::{AtomicBool, Ordering};
+use std::sync::{Arc, OnceLock};
+
+use anyhow::{anyhow, Context};
+use cpal::traits::{DeviceTrait, HostTrait, StreamTrait};
+use cpal::{SampleFormat, SampleRate, StreamConfig};
+use tracing::{info, warn};
+use webrtc_audio_processing::{Config, EchoCancellation, InitializationConfig, NoiseSuppression, Processor};
+
+use crate::audio_ring::AudioRing;
+
+/// 20 ms at 48 kHz, mono — matches the rest of the pipeline and the codec.
+pub const FRAME_SAMPLES: usize = 960;
+/// APM requires strict 10 ms frames at 48 kHz = 480 samples per call.
+const APM_FRAME_SAMPLES: usize = 480;
+const APM_NUM_CHANNELS: usize = 1;
+/// Round-trip delay hint passed to APM; the estimator refines from here.
+/// 60 ms is a reasonable default for CPAL on ALSA / PulseAudio / PipeWire.
+#[allow(dead_code)]
+const STREAM_DELAY_MS: i32 = 60;
+
+// ---------------------------------------------------------------------------
+// Shared APM instance
+// ---------------------------------------------------------------------------
+
+/// Module-level lazily-initialized APM. Shared between capture and playback
+/// so they operate on the same echo-cancellation state — the render frames
+/// pushed by playback are what the capture path subtracts from the mic input.
+static PROCESSOR: OnceLock<Arc<Processor>> = OnceLock::new();
+
+fn get_or_init_processor() -> anyhow::Result<Arc<Processor>> {
+    if let Some(p) = PROCESSOR.get() {
+        return Ok(p.clone());
+    }
+    let init_config = InitializationConfig {
+        num_capture_channels: APM_NUM_CHANNELS as i32,
+        num_render_channels: APM_NUM_CHANNELS as i32,
+        ..Default::default()
+    };
+    let mut processor = Processor::new(&init_config)
+        .map_err(|e| anyhow!("webrtc APM init failed: {e:?}"))?;
+
+    let config = Config {
+        echo_cancellation: Some(EchoCancellation {
+            suppression_level: webrtc_audio_processing::EchoCancellationSuppressionLevel::High,
+            stream_delay_ms: Some(STREAM_DELAY_MS),
+            enable_delay_agnostic: true,
+            enable_extended_filter: true,
+        }),
+        noise_suppression: Some(NoiseSuppression {
+            suppression_level:
+                webrtc_audio_processing::NoiseSuppressionLevel::High,
+        }),
+        enable_high_pass_filter: true,
+        // AGC left off for now — it can fight the Opus encoder's own gain
+        // staging and the adaptive-quality controller. Add later if users
+        // report low mic levels.
+        ..Default::default()
+    };
+    processor.set_config(config);
+
+    let arc = Arc::new(processor);
+    let _ = PROCESSOR.set(arc.clone());
+    info!(
+        stream_delay_ms = STREAM_DELAY_MS,
+        "webrtc APM initialized (AEC3 High + NS High + HPF, AGC off)"
+    );
+    Ok(arc)
+}
+
+// ---------------------------------------------------------------------------
+// Helpers: i16 ↔ f32 and APM frame processing
+// ---------------------------------------------------------------------------
+
+#[inline]
+fn i16_to_f32(s: i16) -> f32 {
+    s as f32 / 32768.0
+}
+
+#[inline]
+fn f32_to_i16(s: f32) -> i16 {
+    (s.clamp(-1.0, 1.0) * 32767.0) as i16
+}
+
+/// Feed a 20 ms (960-sample) playback frame to APM as the render reference.
+/// Splits into two 10 ms halves because APM is strict about frame size.
+fn push_render_frame_20ms(apm: &Processor, pcm: &[i16]) {
+    debug_assert_eq!(pcm.len(), FRAME_SAMPLES);
+    let mut buf = [0f32; APM_FRAME_SAMPLES];
+    for half in pcm.chunks_exact(APM_FRAME_SAMPLES) {
+        for (i, &s) in half.iter().enumerate() {
+            buf[i] = i16_to_f32(s);
+        }
+        // process_render_frame mutates in place. For render we only care
+        // about feeding APM the reference — we discard the output.
+        if let Err(e) = apm.process_render_frame(&mut buf) {
+            warn!("webrtc APM process_render_frame failed: {e:?}");
+        }
+    }
+}
+
+/// Run a 20 ms (960-sample) capture frame through APM's echo cancellation
+/// in place. Splits into two 10 ms halves, runs APM on each, stitches
+/// results back into the caller's buffer.
+fn process_capture_frame_20ms(apm: &Processor, pcm: &mut [i16]) {
+    debug_assert_eq!(pcm.len(), FRAME_SAMPLES);
+    let mut buf = [0f32; APM_FRAME_SAMPLES];
+    for half in pcm.chunks_exact_mut(APM_FRAME_SAMPLES) {
+        for (i, &s) in half.iter().enumerate() {
+            buf[i] = i16_to_f32(s);
+        }
+        if let Err(e) = apm.process_capture_frame(&mut buf) {
+            warn!("webrtc APM process_capture_frame failed: {e:?}");
+        }
+        for (i, d) in half.iter_mut().enumerate() {
+            *d = f32_to_i16(buf[i]);
+        }
+    }
+}
+
+// ---------------------------------------------------------------------------
+// LinuxAecCapture — CPAL mic + WebRTC AEC capture-side processing
+// ---------------------------------------------------------------------------
+
+/// Microphone capture with WebRTC AEC3 applied in place before the codec
+/// sees the samples. Mirrors the public API of `audio_io::AudioCapture` so
+/// downstream code doesn't change.
+pub struct LinuxAecCapture {
+    ring: Arc<AudioRing>,
+    running: Arc<AtomicBool>,
+}
+
+impl LinuxAecCapture {
+    pub fn start() -> Result<Self, anyhow::Error> {
+        // Eagerly init the APM so the playback side can find it already
+        // configured, and so init errors surface on the caller thread
+        // instead of silently failing inside the capture thread.
+        let apm = get_or_init_processor()?;
+
+        let ring = Arc::new(AudioRing::new());
+        let running = Arc::new(AtomicBool::new(true));
+
+        let (init_tx, init_rx) = std::sync::mpsc::sync_channel::<Result<(), String>>(1);
+
+        let ring_cb = ring.clone();
+        let running_clone = running.clone();
+        let apm_capture = apm.clone();
+
+        std::thread::Builder::new()
+            .name("wzp-audio-capture-linuxaec".into())
+            .spawn(move || {
+                let result = (|| -> Result<(), anyhow::Error> {
+                    let host = cpal::default_host();
+                    let device = host
+                        .default_input_device()
+                        .ok_or_else(|| anyhow!("no default input audio device found"))?;
+                    info!(device = %device.name().unwrap_or_default(), "LinuxAEC: using input device");
+
+                    let config = StreamConfig {
+                        channels: 1,
+                        sample_rate: SampleRate(48_000),
+                        buffer_size: cpal::BufferSize::Default,
+                    };
+
+                    let use_f32 = !supports_i16_input(&device)?;
+
+                    let err_cb = |e: cpal::StreamError| {
+                        warn!("LinuxAEC input stream error: {e}");
+                    };
+
+                    // Leftover buffer for when CPAL gives us partial frames.
+                    // We need exactly 960-sample chunks to feed APM.
+                    let leftover = std::sync::Mutex::new(Vec::<i16>::with_capacity(FRAME_SAMPLES * 4));
+
+                    let stream = if use_f32 {
+                        let ring = ring_cb.clone();
+                        let running = running_clone.clone();
+                        let apm = apm_capture.clone();
+                        device.build_input_stream(
+                            &config,
+                            move |data: &[f32], _: &cpal::InputCallbackInfo| {
+                                if !running.load(Ordering::Relaxed) {
+                                    return;
+                                }
+                                let mut lv = leftover.lock().unwrap();
+                                lv.reserve(data.len());
+                                for &s in data {
+                                    lv.push(f32_to_i16(s));
+                                }
+                                drain_frames_through_apm(&mut lv, &apm, &ring);
+                            },
+                            err_cb,
+                            None,
+                        )?
+                    } else {
+                        let ring = ring_cb.clone();
+                        let running = running_clone.clone();
+                        let apm = apm_capture.clone();
+                        device.build_input_stream(
+                            &config,
+                            move |data: &[i16], _: &cpal::InputCallbackInfo| {
+                                if !running.load(Ordering::Relaxed) {
+                                    return;
+                                }
+                                let mut lv = leftover.lock().unwrap();
+                                lv.extend_from_slice(data);
+                                drain_frames_through_apm(&mut lv, &apm, &ring);
+                            },
+                            err_cb,
+                            None,
+                        )?
+                    };
+
+                    stream.play().context("failed to start LinuxAEC input stream")?;
+                    let _ = init_tx.send(Ok(()));
+                    info!("LinuxAEC capture started (AEC3 active)");
+
+                    while running_clone.load(Ordering::Relaxed) {
+                        std::thread::park_timeout(std::time::Duration::from_millis(200));
+                    }
+                    drop(stream);
+                    Ok(())
+                })();
+
+                if let Err(e) = result {
+                    let _ = init_tx.send(Err(e.to_string()));
+                }
+            })?;
+
+        init_rx
+            .recv()
+            .map_err(|_| anyhow!("LinuxAEC capture thread exited before signaling"))?
+            .map_err(|e| anyhow!("{e}"))?;
+
+        Ok(Self { ring, running })
+    }
+
+    pub fn ring(&self) -> &Arc<AudioRing> {
+        &self.ring
+    }
+
+    pub fn stop(&self) {
+        self.running.store(false, Ordering::Relaxed);
+    }
+}
+
+impl Drop for LinuxAecCapture {
+    fn drop(&mut self) {
+        self.stop();
+    }
+}
+
+/// Pull whole 960-sample frames out of the leftover buffer, run them through
+/// APM's capture-side processing, and push to the ring. Leaves any partial
+/// sub-960 remainder in `leftover` for the next callback.
+fn drain_frames_through_apm(leftover: &mut Vec<i16>, apm: &Processor, ring: &AudioRing) {
+    let mut frame = [0i16; FRAME_SAMPLES];
+    while leftover.len() >= FRAME_SAMPLES {
+        frame.copy_from_slice(&leftover[..FRAME_SAMPLES]);
+        process_capture_frame_20ms(apm, &mut frame);
+        ring.write(&frame);
+        leftover.drain(..FRAME_SAMPLES);
+    }
+}
+
+// ---------------------------------------------------------------------------
+// LinuxAecPlayback — CPAL speaker output + WebRTC AEC render-side tee
+// ---------------------------------------------------------------------------
+
+/// Speaker playback with a render-side tee: each frame written to CPAL is
+/// ALSO fed to APM via `process_render_frame` as the echo-cancellation
+/// reference signal. This is the "tee the playback ring" approach (Zoom,
+/// Teams, Jitsi) — deterministic, does not depend on PulseAudio loopback or
+/// PipeWire monitor sources.
+pub struct LinuxAecPlayback {
+    ring: Arc<AudioRing>,
+    running: Arc<AtomicBool>,
+}
+
+impl LinuxAecPlayback {
+    pub fn start() -> Result<Self, anyhow::Error> {
+        let apm = get_or_init_processor()?;
+
+        let ring = Arc::new(AudioRing::new());
+        let running = Arc::new(AtomicBool::new(true));
+
+        let (init_tx, init_rx) = std::sync::mpsc::sync_channel::<Result<(), String>>(1);
+
+        let ring_cb = ring.clone();
+        let running_clone = running.clone();
+        let apm_render = apm.clone();
+
+        std::thread::Builder::new()
+            .name("wzp-audio-playback-linuxaec".into())
+            .spawn(move || {
+                let result = (|| -> Result<(), anyhow::Error> {
+                    let host = cpal::default_host();
+                    let device = host
+                        .default_output_device()
+                        .ok_or_else(|| anyhow!("no default output audio device found"))?;
+                    info!(device = %device.name().unwrap_or_default(), "LinuxAEC: using output device");
+
+                    let config = StreamConfig {
+                        channels: 1,
+                        sample_rate: SampleRate(48_000),
+                        buffer_size: cpal::BufferSize::Default,
+                    };
+
+                    let use_f32 = !supports_i16_output(&device)?;
+
+                    let err_cb = |e: cpal::StreamError| {
+                        warn!("LinuxAEC output stream error: {e}");
+                    };
+
+                    // Same 960-sample batching approach as the capture side:
+                    // CPAL may ask for N samples in a callback where N doesn't
+                    // divide 960. We accumulate partial frames in a Vec and
+                    // feed APM as soon as we have a whole 20 ms frame.
+                    let carry = std::sync::Mutex::new(Vec::<i16>::with_capacity(FRAME_SAMPLES * 4));
+
+                    let stream = if use_f32 {
+                        let ring = ring_cb.clone();
+                        let apm = apm_render.clone();
+                        device.build_output_stream(
+                            &config,
+                            move |data: &mut [f32], _: &cpal::OutputCallbackInfo| {
+                                fill_output_and_tee_f32(data, &ring, &apm, &carry);
+                            },
+                            err_cb,
+                            None,
+                        )?
+                    } else {
+                        let ring = ring_cb.clone();
+                        let apm = apm_render.clone();
+                        device.build_output_stream(
+                            &config,
+                            move |data: &mut [i16], _: &cpal::OutputCallbackInfo| {
+                                fill_output_and_tee_i16(data, &ring, &apm, &carry);
+                            },
+                            err_cb,
+                            None,
+                        )?
+                    };
+
+                    stream.play().context("failed to start LinuxAEC output stream")?;
+                    let _ = init_tx.send(Ok(()));
+                    info!("LinuxAEC playback started (render tee active)");
+
+                    while running_clone.load(Ordering::Relaxed) {
+                        std::thread::park_timeout(std::time::Duration::from_millis(200));
+                    }
+                    drop(stream);
+                    Ok(())
+                })();
+
+                if let Err(e) = result {
+                    let _ = init_tx.send(Err(e.to_string()));
+                }
+            })?;
+
+        init_rx
+            .recv()
+            .map_err(|_| anyhow!("LinuxAEC playback thread exited before signaling"))?
+            .map_err(|e| anyhow!("{e}"))?;
+
+        Ok(Self { ring, running })
+    }
+
+    pub fn ring(&self) -> &Arc<AudioRing> {
+        &self.ring
+    }
+
+    pub fn stop(&self) {
+        self.running.store(false, Ordering::Relaxed);
+    }
+}
+
+impl Drop for LinuxAecPlayback {
+    fn drop(&mut self) {
+        self.stop();
+    }
+}
+
+fn fill_output_and_tee_i16(
+    data: &mut [i16],
+    ring: &AudioRing,
+    apm: &Processor,
+    carry: &std::sync::Mutex<Vec<i16>>,
+) {
+    let read = ring.read(data);
+    for s in &mut data[read..] {
+        *s = 0;
+    }
+    tee_render_samples(data, apm, carry);
+}
+
+fn fill_output_and_tee_f32(
+    data: &mut [f32],
+    ring: &AudioRing,
+    apm: &Processor,
+    carry: &std::sync::Mutex<Vec<i16>>,
+) {
+    let mut tmp = vec![0i16; data.len()];
+    let read = ring.read(&mut tmp);
+    for s in &mut tmp[read..] {
+        *s = 0;
+    }
+    for (d, &s) in data.iter_mut().zip(tmp.iter()) {
+        *d = i16_to_f32(s);
+    }
+    tee_render_samples(&tmp, apm, carry);
+}
+
+/// Push CPAL-bound samples into APM's render-side input for echo cancellation.
+/// Uses a carry buffer to batch into exact 960-sample (20 ms) frames.
+fn tee_render_samples(samples: &[i16], apm: &Processor, carry: &std::sync::Mutex<Vec<i16>>) {
+    let mut lv = carry.lock().unwrap();
+    lv.extend_from_slice(samples);
+    while lv.len() >= FRAME_SAMPLES {
+        let mut frame = [0i16; FRAME_SAMPLES];
+        frame.copy_from_slice(&lv[..FRAME_SAMPLES]);
+        push_render_frame_20ms(apm, &frame);
+        lv.drain(..FRAME_SAMPLES);
+    }
+}
+
+// ---------------------------------------------------------------------------
+// CPAL format helpers (duplicated from audio_io.rs to keep the modules
+// independent — each backend file is a self-contained unit)
+// ---------------------------------------------------------------------------
+
+fn supports_i16_input(device: &cpal::Device) -> Result<bool, anyhow::Error> {
+    let supported = device
+        .supported_input_configs()
+        .context("failed to query input configs")?;
+    for cfg in supported {
+        if cfg.sample_format() == SampleFormat::I16
+            && cfg.min_sample_rate() <= SampleRate(48_000)
+            && cfg.max_sample_rate() >= SampleRate(48_000)
+            && cfg.channels() >= 1
+        {
+            return Ok(true);
+        }
+    }
+    Ok(false)
+}
+
+fn supports_i16_output(device: &cpal::Device) -> Result<bool, anyhow::Error> {
+    let supported = device
+        .supported_output_configs()
+        .context("failed to query output configs")?;
+    for cfg in supported {
+        if cfg.sample_format() == SampleFormat::I16
+            && cfg.min_sample_rate() <= SampleRate(48_000)
+            && cfg.max_sample_rate() >= SampleRate(48_000)
+            && cfg.channels() >= 1
+        {
+            return Ok(true);
+        }
+    }
+    Ok(false)
+}