fix(audio): bound playback handoff queue

Fixes the two P2 efficacy findings from the Codex audit of the W12
profiles feature.

FEC was enabled on the encoder but never used: the jitter buffer's
loss path did pure PLC, so the redundancy was wasted bitrate. Now the
gap path reconstructs the lost frame from the next buffered packet via
Opus in-band FEC (new `AudioDecoder::decode_fec`, libopus decode with
fec=true into a one-frame buffer), keeping that packet for its own
normal decode and falling back to PLC if FEC decode fails. This is the
documented libopus FEC pattern; receiver-side only, no wire change.

DTX was enabled on BadNetwork but provided no benefit — the capture
noise gate already suppresses silence transmission, and the broadcast
DTX silence packets only created seq gaps that grew the jitter cushion.
All profiles now set dtx=false (plumbing kept for a future revisit).

Adds a jitter-buffer test proving FEC reconstruction beats pure PLC
(RMS error < 0.75x) and that the FEC source packet stays buffered.
500 lib tests, clippy + fmt clean, release build clean.

Co-Authored-By: Codex (gpt-5.5) <noreply@openai.com>
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>

.gitea/workflows/ci.yml

@@ -0,0 +1,42 @@
+name: CI
+
+# Runs on the self-hosted host-mode runner on the desktop (label `arch`). The
+# gitbutter VPS only queues the job; all compile/test compute happens locally.
+on:
+  push:
+    branches: [main]
+  pull_request:
+  workflow_dispatch:
+
+jobs:
+  check:
+    runs-on: arch
+    steps:
+      - name: Checkout
+        uses: actions/checkout@v4
+
+      - name: Toolchain versions
+        run: |
+          rustc --version
+          cargo --version
+          cargo clippy --version
+          cargo deny --version
+          cargo audit --version
+
+      - name: Format check
+        run: cargo fmt --all -- --check
+
+      - name: Clippy (all targets, warnings as errors)
+        run: cargo clippy --all-targets -- -D warnings
+
+      - name: Tests
+        run: cargo test --all-targets
+
+      - name: Doc tests
+        run: cargo test --doc
+
+      - name: cargo-deny (advisories, bans, licenses, sources)
+        run: cargo deny check
+
+      - name: cargo-audit
+        run: cargo audit

Cargo.lock

@@ -200,9 +200,9 @@ checksum = "940b3a0ca603d1eade50a4846a2afffd5ef57a9feac2c0e2ec2e14f9ead76000"
 
 [[package]]
 name = "anyhow"
-version = "1.0.102"
+version = "1.0.103"
 source = "registry+https://github.com/rust-lang/crates.io-index"
-checksum = "7f202df86484c868dbad7eaa557ef785d5c66295e41b460ef922eca0723b842c"
+checksum = "2a4385e2e34eb35d6b3efe798b9eb88096925d87726c0798709bf56d9ed84af3"
 
 [[package]]
 name = "arbitrary"
@@ -3682,9 +3682,9 @@ checksum = "6b947ae49db0d222b1dbc6b113ce7248a3fc3a6ca21b696717bfc000ba4484d8"
 
 [[package]]
 name = "memmap2"
-version = "0.9.10"
+version = "0.9.11"
 source = "registry+https://github.com/rust-lang/crates.io-index"
-checksum = "714098028fe011992e1c3962653c96b2d578c4b4bce9036e15ff220319b1e0e3"
+checksum = "d1219ed1b7f229ee7104d281dd01d6802fe28bb6e95d292942c4daacdeb798c0"
 dependencies = [
  "libc",
 ]

build.rs

@@ -0,0 +1,22 @@
+use std::process::Command;
+
+fn main() {
+    println!("cargo:rerun-if-changed=.git/HEAD");
+    if let Ok(head) = std::fs::read_to_string(".git/HEAD")
+        && let Some(reference) = head.strip_prefix("ref: ")
+    {
+        println!("cargo:rerun-if-changed=.git/{}", reference.trim());
+    }
+
+    let short = Command::new("git")
+        .args(["rev-parse", "--short=8", "HEAD"])
+        .output()
+        .ok()
+        .filter(|output| output.status.success())
+        .and_then(|output| String::from_utf8(output.stdout).ok())
+        .map(|value| value.trim().to_string())
+        .filter(|value| !value.is_empty())
+        .unwrap_or_else(|| "unknown".to_string());
+
+    println!("cargo:rustc-env=PEERSPEAK_GIT_SHORT={short}");
+}

packaging/PKGBUILD

@@ -1,7 +1,7 @@
 # Maintainer: mollusk <jitty+lc1iz0dc@protonmail.com>
 pkgname=peerspeak-git
 _pkgname=peerspeak
-pkgver=0.5.0.r0.g0000000
+pkgver=0.6.1.r310.g660261a
 pkgrel=1
 pkgdesc="Decentralized peer-to-peer voice chat (Rust/iroh/PipeWire/Opus/iced)"
 arch=('x86_64')

src/app/mod.rs

@@ -4,7 +4,7 @@ use crate::audio::clip_player::{
 };
 use crate::audio::eq::{EQ_GAIN_DB_MAX, EQ_GAIN_DB_MIN, EqSettings};
 use crate::audio::{AudioDevice, enumerate_audio_devices};
-use crate::config::{AppConfig, NetworkMode, RecordingMode, RoomLayout};
+use crate::config::{AppConfig, AudioProfile, NetworkMode, RecordingMode, RoomLayout};
 use crate::core::{
     CoreController,
     messages::{CoreCommand, UiEvent},
@@ -34,6 +34,17 @@ use tokio::sync::Mutex;
 
 static UI_RX: OnceLock<Mutex<Option<tokio::sync::mpsc::Receiver<UiEvent>>>> = OnceLock::new();
 
+const APP_VERSION: &str = env!("CARGO_PKG_VERSION");
+const GIT_SHORT: &str = env!("PEERSPEAK_GIT_SHORT");
+
+fn app_build_label() -> String {
+    if GIT_SHORT == "unknown" {
+        format!("PeerSpeak v{APP_VERSION}")
+    } else {
+        format!("PeerSpeak v{APP_VERSION} ({GIT_SHORT})")
+    }
+}
+
 #[derive(Debug, Clone, Copy, PartialEq, Eq)]
 pub enum Screen {
     Home,
@@ -366,6 +377,7 @@ pub enum AppMessage {
     /// immediately but does not persist (saved once on release via NoiseGateChanged).
     NoiseGateDragging(f32),
     NetworkModeSelected(NetworkMode),
+    AudioProfileSelected(AudioProfile),
     RecordingModeSelected(RecordingMode),
     /// Choose the friends presence posture (W7): invisible / normal / discoverable.
     PresenceModeSelected(PresenceMode),
@@ -908,6 +920,7 @@ impl Default for AppState {
         let _ = controller.send(CoreCommand::SetInputVolume(config.input_volume));
         let _ = controller.send(CoreCommand::SetOutputVolume(config.output_volume));
         let _ = controller.send(CoreCommand::SetNetworkMode(config.network_mode));
+        let _ = controller.send(CoreCommand::SetAudioProfile(config.audio_profile));
         let _ = controller.send(CoreCommand::SetRecordingMode(config.recording_mode));
         let _ = controller.send(CoreCommand::SetPixelpassPath(config.pixelpass_path.clone()));
         let _ = controller.send(CoreCommand::SetPresenceMode(config.presence_mode));
@@ -1100,6 +1113,7 @@ fn effective_background_bytes(
 }
 
 pub fn run_gui() -> iced::Result {
+    crate::log_msg(&format!("{} starting", app_build_label()));
     // Restore the last window size (saved on close). Position is restored too,
     // but only on X11 — Wayland's xdg-shell gives clients no way to set their own
     // position, so we center there and leave placement to the compositor.
@@ -2038,6 +2052,12 @@ fn update(state: &mut AppState, message: AppMessage) -> Task<AppMessage> {
             // Applied on the next join, since the endpoint is rebuilt then.
             let _ = state.controller.send(CoreCommand::SetNetworkMode(mode));
         }
+        AppMessage::AudioProfileSelected(profile) => {
+            state.config.audio_profile = profile;
+            state.config.save();
+            // Applies live to the running encoder, and to the next call.
+            let _ = state.controller.send(CoreCommand::SetAudioProfile(profile));
+        }
         AppMessage::RecordingModeSelected(mode) => {
             state.config.recording_mode = mode;
             state.config.save();
@@ -3111,6 +3131,19 @@ fn network_mode_hint(mode: NetworkMode) -> &'static str {
     }
 }
 
+/// One-line explanation of an audio/network profile for the settings picker (W12).
+fn audio_profile_hint(profile: AudioProfile) -> &'static str {
+    match profile {
+        AudioProfile::LowLatency => {
+            "Lowest delay, no loss recovery. Best on a clean LAN or wired link."
+        }
+        AudioProfile::Balanced => "Default: voice quality with light loss recovery.",
+        AudioProfile::BadNetwork => {
+            "Most resilient on a lossy/congested link: heavier loss recovery, lower bitrate."
+        }
+    }
+}
+
 /// One-line explanation of a recording mode for the settings picker.
 fn recording_mode_hint(mode: RecordingMode) -> &'static str {
     match mode {
@@ -3806,6 +3839,7 @@ fn connect_card(state: &AppState) -> Element<'_, AppMessage> {
     let subtitle = text("NAT-traversing full-mesh voice chat")
         .size(16)
         .color(color_subtext);
+    let build_label = text(app_build_label()).size(11).color(color_subtext);
 
     let nickname_input = column![
         text("Nickname").size(14).color(color_subtext),
@@ -3861,6 +3895,7 @@ fn connect_card(state: &AppState) -> Element<'_, AppMessage> {
         column![
             logo,
             subtitle,
+            build_label,
             vertical_space(20.0),
             nickname_input,
             vertical_space(16.0),
@@ -4967,6 +5002,17 @@ fn view(state: &AppState) -> Element<'_, AppMessage> {
                         control
                     },
                 ].spacing(8).width(iced::Length::Fill),
+                vertical_space(section_gap),
+                section_header("Connection quality"),
+                column![
+                    pick_list(
+                        &AudioProfile::ALL[..],
+                        Some(state.config.audio_profile),
+                        AppMessage::AudioProfileSelected,
+                    ).width(iced::Length::Fill),
+                    text(audio_profile_hint(state.config.audio_profile)).size(11).color(color_subtext),
+                    text("Applies immediately, even mid-call.").size(11).color(color_subtext),
+                ].spacing(4).width(iced::Length::Fill),
             ]
             .spacing(10)
             .width(iced::Length::Fill)
@@ -5240,6 +5286,9 @@ fn view(state: &AppState) -> Element<'_, AppMessage> {
         for category in SettingsCategory::ALL {
             settings_nav = settings_nav.push(category_button(category));
         }
+        settings_nav = settings_nav
+            .push(iced::widget::Space::new().height(iced::Length::Fill))
+            .push(text(app_build_label()).size(11).color(color_subtext));
         let settings_nav = container(settings_nav)
             .padding(12)
             .width(iced::Length::Fixed(220.0))
@@ -5258,6 +5307,8 @@ fn view(state: &AppState) -> Element<'_, AppMessage> {
                     .width(iced::Length::Fill),
                     vertical_space(10.0),
                     settings_body,
+                    vertical_space(10.0),
+                    text(app_build_label()).size(11).color(color_subtext),
                 ]
                 .spacing(8)
                 .width(iced::Length::Fill),

src/audio/multitrack.rs

@@ -12,9 +12,11 @@
 //! This module is pure plumbing over [`WavWriter`]: no audio decode, no
 //! networking, no realtime work. The mixer (a non-RT task) drives it.
 
-use std::collections::{HashMap, VecDeque};
+use std::collections::{HashMap, HashSet, VecDeque};
 use std::io;
 use std::path::{Path, PathBuf};
+use std::sync::mpsc::{self, SyncSender, TrySendError};
+use std::thread::{self, JoinHandle};
 
 use iroh::EndpointId;
 
@@ -24,6 +26,8 @@ use crate::core::jitter::FRAME_SAMPLES;
 /// Cap on the silence chunk written at once when pre-padding a late joiner, so a
 /// long-running call can't trigger a single multi-hundred-MB allocation.
 const SILENCE_CHUNK: usize = FRAME_SAMPLES * 256;
+const WRITER_QUEUE_CYCLES: usize = 256;
+const DROP_LOG_INTERVAL_CYCLES: u64 = 256;
 
 /// Cap on buffered mic samples (~200ms @ 48kHz). Bounds how far the mic track
 /// can drift if the capture clock runs ahead of the mixer cycle; past it the
@@ -56,41 +60,6 @@ pub fn create_session_dir(base: &Path, now_unix_secs: u64) -> io::Result<PathBuf
     ))
 }
 
-/// One output track: its WAV writer plus whether it has been written *this*
-/// cycle (so `end_cycle` knows which tracks to pad with silence).
-struct Track {
-    writer: WavWriter,
-    written_this_cycle: bool,
-}
-
-impl Track {
-    fn create(path: &Path) -> io::Result<Self> {
-        Ok(Self {
-            writer: WavWriter::new(path)?,
-            written_this_cycle: false,
-        })
-    }
-
-    /// Append `frame` fitted to exactly `frame_samples` (zero-padded if short),
-    /// and mark the track as written for this cycle.
-    fn write_frame(&mut self, frame: &[i16], frame_samples: usize) -> io::Result<()> {
-        self.writer.write_samples(&fit(frame, frame_samples))?;
-        self.written_this_cycle = true;
-        Ok(())
-    }
-
-    /// Append `samples` of silence (no cycle-marking — used for padding).
-    fn write_silence(&mut self, samples: usize) -> io::Result<()> {
-        let mut remaining = samples;
-        while remaining > 0 {
-            let n = remaining.min(SILENCE_CHUNK);
-            self.writer.write_samples(&vec![0i16; n])?;
-            remaining -= n;
-        }
-        Ok(())
-    }
-}
-
 /// Return `frame` resized to exactly `n` samples: truncated if longer (shouldn't
 /// happen — Opus frames are uniform), zero-padded if shorter.
 fn fit(frame: &[i16], n: usize) -> Vec<i16> {
@@ -126,41 +95,210 @@ pub fn track_filename(name: &str, id: &EndpointId) -> String {
     format!("{slug}-{short}.wav")
 }
 
-/// A live multitrack recording: per-peer stems + your mic, plus an optional
+#[derive(Default)]
-/// mixed track, all under one session directory and clocked together.
+struct PendingCycle {
-pub struct MultitrackRecorder {
+    new_peers: Vec<NewPeer>,
-    dir: PathBuf,
+    peer_frames: HashMap<EndpointId, Vec<i16>>,
-    frame_samples: usize,
+    mix_frame: Option<Vec<i16>>,
-    /// Cycles recorded so far = the shared length (in frames) of every track.
-    cycles: u64,
-    peers: HashMap<EndpointId, Track>,
-    /// Your mic track. Fed asynchronously from the capture thread via
-    /// [`push_mic`](MultitrackRecorder::push_mic) into `mic_fifo`, then drained
-    /// one frame per `end_cycle` so it aligns with the cycle clock.
-    mic: WavWriter,
-    mic_fifo: VecDeque<i16>,
-    /// Present in "Both" mode (stems + mixed), absent in "stems only".
-    mix: Option<Track>,
 }
 
-impl MultitrackRecorder {
+struct NewPeer {
-    /// Create a recording in `dir` (which must already exist). `with_mix` adds
+    id: EndpointId,
-    /// the convenience mixed track (`mix.wav`). Your mic is always `me.wav`.
+    filename: String,
-    pub fn create(dir: &Path, frame_samples: usize, with_mix: bool) -> io::Result<Self> {
+}
+
+struct CycleBatch {
+    new_peers: Vec<NewPeer>,
+    mic_frame: Vec<i16>,
+    mix_frame: Option<Vec<i16>>,
+    peer_frames: HashMap<EndpointId, Vec<i16>>,
+}
+
+trait SampleWriter {
+    fn write_samples(&mut self, samples: &[i16]) -> io::Result<()>;
+    fn finalize(self) -> io::Result<()>;
+}
+
+impl SampleWriter for WavWriter {
+    fn write_samples(&mut self, samples: &[i16]) -> io::Result<()> {
+        WavWriter::write_samples(self, samples)
+    }
+
+    fn finalize(self) -> io::Result<()> {
+        WavWriter::finalize(self)
+    }
+}
+
+struct WriterState<W> {
+    dir: PathBuf,
+    frame_samples: usize,
+    peers: HashMap<EndpointId, W>,
+    mic: W,
+    mix: Option<W>,
+    cycles_written: u64,
+}
+
+impl WriterState<WavWriter> {
+    fn create(dir: &Path, frame_samples: usize, with_mix: bool) -> io::Result<Self> {
         let mic = WavWriter::new(&dir.join("me.wav"))?;
         let mix = if with_mix {
-            Some(Track::create(&dir.join("mix.wav"))?)
+            Some(WavWriter::new(&dir.join("mix.wav"))?)
         } else {
             None
         };
         Ok(Self {
             dir: dir.to_path_buf(),
             frame_samples,
-            cycles: 0,
             peers: HashMap::new(),
             mic,
-            mic_fifo: VecDeque::new(),
             mix,
+            cycles_written: 0,
+        })
+    }
+}
+
+impl<W: SampleWriter> WriterState<W> {
+    fn apply_batch<F>(&mut self, batch: &CycleBatch, mut create_peer: F) -> io::Result<()>
+    where
+        F: FnMut(&Path) -> io::Result<W>,
+    {
+        for peer in &batch.new_peers {
+            if !self.peers.contains_key(&peer.id) {
+                let writer = create_peer(&self.dir.join(&peer.filename))?;
+                self.peers.insert(peer.id, writer);
+                let pad = self.back_pad_samples()?;
+                let writer = self.peers.get_mut(&peer.id).unwrap();
+                Self::write_silence(writer, pad)?;
+            }
+        }
+
+        self.mic.write_samples(&batch.mic_frame)?;
+        if let Some(mix) = self.mix.as_mut() {
+            if let Some(frame) = batch.mix_frame.as_deref() {
+                mix.write_samples(frame)?;
+            } else {
+                Self::write_silence(mix, self.frame_samples)?;
+            }
+        }
+
+        let silence = vec![0i16; self.frame_samples];
+        for (id, writer) in &mut self.peers {
+            let frame = batch
+                .peer_frames
+                .get(id)
+                .map(Vec::as_slice)
+                .unwrap_or(&silence);
+            writer.write_samples(frame)?;
+        }
+        self.cycles_written += 1;
+        Ok(())
+    }
+
+    fn back_pad_samples(&self) -> io::Result<usize> {
+        let cycles = usize::try_from(self.cycles_written)
+            .map_err(|_| io::Error::other("multitrack recording too long"))?;
+        cycles
+            .checked_mul(self.frame_samples)
+            .ok_or_else(|| io::Error::other("multitrack recording too long"))
+    }
+
+    fn write_silence(writer: &mut W, samples: usize) -> io::Result<()> {
+        let mut remaining = samples;
+        let silence = vec![0i16; remaining.min(SILENCE_CHUNK)];
+        while remaining > 0 {
+            let n = remaining.min(silence.len());
+            writer.write_samples(&silence[..n])?;
+            remaining -= n;
+        }
+        Ok(())
+    }
+
+    fn finalize(self) -> io::Result<()> {
+        let mut first_finalize_error = None;
+        record_first_error(&mut first_finalize_error, self.mic.finalize());
+        if let Some(mix) = self.mix {
+            record_first_error(&mut first_finalize_error, mix.finalize());
+        }
+        for writer in self.peers.into_values() {
+            record_first_error(&mut first_finalize_error, writer.finalize());
+        }
+        if let Some(e) = first_finalize_error {
+            Err(e)
+        } else {
+            Ok(())
+        }
+    }
+}
+
+fn record_first_error(slot: &mut Option<io::Error>, result: io::Result<()>) {
+    if slot.is_none()
+        && let Err(e) = result
+    {
+        *slot = Some(e);
+    }
+}
+
+/// Applies whole-cycle batches on the writer thread. Each applied batch appends
+/// exactly `frame_samples` to every existing track, and a dropped batch never
+/// reaches this loop for any track, so stem lengths stay equal even when the
+/// bounded queue applies back-pressure.
+fn writer_thread_main(
+    mut state: WriterState<WavWriter>,
+    batch_rx: mpsc::Receiver<CycleBatch>,
+) -> io::Result<()> {
+    let mut first_write_error = None;
+
+    for batch in batch_rx {
+        if first_write_error.is_none()
+            && let Err(e) = state.apply_batch(&batch, WavWriter::new)
+        {
+            first_write_error = Some(e);
+        }
+    }
+
+    let finalize_result = state.finalize();
+    if let Some(e) = first_write_error {
+        Err(e)
+    } else {
+        finalize_result
+    }
+}
+
+/// A live multitrack recording: per-peer stems + your mic, plus an optional
+/// mixed track, all under one session directory and clocked together.
+pub struct MultitrackRecorder {
+    dir: PathBuf,
+    frame_samples: usize,
+    known_peers: HashSet<EndpointId>,
+    /// Your mic track. Fed asynchronously from the capture thread via
+    /// [`push_mic`](MultitrackRecorder::push_mic) into `mic_fifo`, then drained
+    /// one frame per `end_cycle` so it aligns with the cycle clock.
+    mic_fifo: VecDeque<i16>,
+    /// Present in "Both" mode (stems + mixed), absent in "stems only".
+    with_mix: bool,
+    batch_tx: SyncSender<CycleBatch>,
+    writer_thread: JoinHandle<io::Result<()>>,
+    dropped_cycles: u64,
+    pending: PendingCycle,
+}
+
+impl MultitrackRecorder {
+    /// Create a recording in `dir` (which must already exist). `with_mix` adds
+    /// the convenience mixed track (`mix.wav`). Your mic is always `me.wav`.
+    pub fn create(dir: &Path, frame_samples: usize, with_mix: bool) -> io::Result<Self> {
+        let writer_state = WriterState::create(dir, frame_samples, with_mix)?;
+        let (batch_tx, batch_rx) = mpsc::sync_channel(WRITER_QUEUE_CYCLES);
+        let writer_thread = thread::spawn(move || writer_thread_main(writer_state, batch_rx));
+        Ok(Self {
+            dir: dir.to_path_buf(),
+            frame_samples,
+            known_peers: HashSet::new(),
+            mic_fifo: VecDeque::new(),
+            with_mix,
+            batch_tx,
+            writer_thread,
+            dropped_cycles: 0,
+            pending: PendingCycle::default(),
         })
     }
 
@@ -173,12 +311,14 @@ impl MultitrackRecorder {
     /// so it aligns with the others. Idempotent: a peer already tracked is left
     /// as-is (re-announce / name change doesn't restart their file).
     pub fn add_peer(&mut self, id: EndpointId, name: &str) -> io::Result<()> {
-        if self.peers.contains_key(&id) {
+        if self.known_peers.contains(&id) {
             return Ok(());
         }
-        let mut track = Track::create(&self.dir.join(track_filename(name, &id)))?;
+        self.known_peers.insert(id);
-        track.write_silence(self.cycles as usize * self.frame_samples)?;
+        self.pending.new_peers.push(NewPeer {
-        self.peers.insert(id, track);
+            id,
+            filename: track_filename(name, &id),
+        });
         Ok(())
     }
 
@@ -186,11 +326,13 @@ impl MultitrackRecorder {
     /// registered yet (write raced ahead of the join event), auto-register it
     /// with an id-only name so no audio is dropped.
     pub fn write_peer(&mut self, id: EndpointId, frame: &[i16]) -> io::Result<()> {
-        if !self.peers.contains_key(&id) {
+        if !self.known_peers.contains(&id) {
             self.add_peer(id, "")?;
         }
-        let fs = self.frame_samples;
+        self.pending
-        self.peers.get_mut(&id).unwrap().write_frame(frame, fs)
+            .peer_frames
+            .insert(id, fit(frame, self.frame_samples));
+        Ok(())
     }
 
     /// Buffer a frame of your transmitted mic audio (called from the capture
@@ -215,9 +357,8 @@ impl MultitrackRecorder {
     /// Record the finished mixed-bus frame for the current cycle (no-op in
     /// stems-only mode).
     pub fn write_mix(&mut self, frame: &[i16]) -> io::Result<()> {
-        let fs = self.frame_samples;
+        if self.with_mix {
-        if let Some(mix) = self.mix.as_mut() {
+            self.pending.mix_frame = Some(fit(frame, self.frame_samples));
-            mix.write_frame(frame, fs)?;
         }
         Ok(())
     }
@@ -230,28 +371,63 @@ impl MultitrackRecorder {
         // Mic: always one frame per cycle, drained from the FIFO (silence on
         // underrun), so it tracks the cycle clock like the peer stems.
         let mic_frame = self.drain_mic(fs);
-        self.mic.write_samples(&mic_frame)?;
+        let mut pending = std::mem::take(&mut self.pending);
-        // Peers + the optional mix track: pad any not written this cycle.
+        pending.new_peers.sort_by(|a, b| {
-        for track in self.peers.values_mut().chain(self.mix.as_mut()) {
+            a.filename
-            if !track.written_this_cycle {
+                .cmp(&b.filename)
-                track.write_silence(fs)?;
+                .then_with(|| a.id.to_string().cmp(&b.id.to_string()))
+        });
+        let batch = CycleBatch {
+            new_peers: pending.new_peers,
+            mic_frame,
+            mix_frame: if self.with_mix {
+                pending.mix_frame
+            } else {
+                None
+            },
+            peer_frames: pending.peer_frames,
+        };
+        match self.batch_tx.try_send(batch) {
+            Ok(()) => Ok(()),
+            Err(TrySendError::Full(batch)) => {
+                for peer in &batch.new_peers {
+                    self.known_peers.remove(&peer.id);
+                }
+                self.dropped_cycles = self.dropped_cycles.saturating_add(1);
+                if self.dropped_cycles == 1
+                    || self.dropped_cycles.is_multiple_of(DROP_LOG_INTERVAL_CYCLES)
+                {
+                    crate::log_msg(&format!(
+                        "multitrack recording: writer queue full; dropped {} cycle(s)",
+                        self.dropped_cycles
+                    ));
+                }
+                Ok(())
             }
-            track.written_this_cycle = false;
+            Err(TrySendError::Disconnected(_)) => Err(io::Error::new(
+                io::ErrorKind::BrokenPipe,
+                "multitrack writer thread stopped",
+            )),
         }
-        self.cycles += 1;
-        Ok(())
     }
 
     /// Finalize every track's WAV header. Consumes the recorder.
     pub fn finalize(self) -> io::Result<()> {
-        self.mic.finalize()?;
+        let Self {
-        if let Some(mix) = self.mix {
+            dir: _,
-            mix.writer.finalize()?;
+            frame_samples: _,
-        }
+            known_peers: _,
-        for (_, track) in self.peers {
+            mic_fifo: _,
-            track.writer.finalize()?;
+            with_mix: _,
-        }
+            batch_tx,
-        Ok(())
+            writer_thread,
+            dropped_cycles: _,
+            pending: _,
+        } = self;
+        drop(batch_tx);
+        writer_thread
+            .join()
+            .unwrap_or_else(|_| Err(io::Error::other("multitrack writer thread panicked")))
     }
 }
 
@@ -277,6 +453,51 @@ mod tests {
         d
     }
 
+    #[derive(Default)]
+    struct TestWriter {
+        samples: Vec<i16>,
+    }
+
+    impl SampleWriter for TestWriter {
+        fn write_samples(&mut self, samples: &[i16]) -> io::Result<()> {
+            self.samples.extend_from_slice(samples);
+            Ok(())
+        }
+
+        fn finalize(self) -> io::Result<()> {
+            Ok(())
+        }
+    }
+
+    fn test_writer_state(frame_samples: usize, with_mix: bool) -> WriterState<TestWriter> {
+        WriterState {
+            dir: PathBuf::new(),
+            frame_samples,
+            peers: HashMap::new(),
+            mic: TestWriter::default(),
+            mix: if with_mix {
+                Some(TestWriter::default())
+            } else {
+                None
+            },
+            cycles_written: 0,
+        }
+    }
+
+    fn test_batch(
+        new_peers: Vec<NewPeer>,
+        mic_frame: Vec<i16>,
+        mix_frame: Option<Vec<i16>>,
+        peer_frames: Vec<(EndpointId, Vec<i16>)>,
+    ) -> CycleBatch {
+        CycleBatch {
+            new_peers,
+            mic_frame,
+            mix_frame,
+            peer_frames: peer_frames.into_iter().collect(),
+        }
+    }
+
     #[test]
     fn fit_pads_and_truncates() {
         assert_eq!(fit(&[1, 2], 4), vec![1, 2, 0, 0]);
@@ -311,6 +532,110 @@ mod tests {
         let _ = std::fs::remove_dir_all(&base);
     }
 
+    #[test]
+    fn apply_batch_advances_existing_tracks_and_back_pads_late_peer() {
+        let frame = 3;
+        let early = an_id();
+        let late = an_id();
+        let mut state = test_writer_state(frame, true);
+        state.cycles_written = 2;
+        state.mic.samples = vec![8; 2 * frame];
+        state.mix.as_mut().unwrap().samples = vec![6; 2 * frame];
+        state.peers.insert(
+            early,
+            TestWriter {
+                samples: vec![1; 2 * frame],
+            },
+        );
+
+        let batch = test_batch(
+            vec![NewPeer {
+                id: late,
+                filename: "late.wav".to_string(),
+            }],
+            vec![9; frame],
+            None,
+            vec![(early, vec![2; frame]), (late, vec![7; frame])],
+        );
+        state
+            .apply_batch(&batch, |_| Ok(TestWriter::default()))
+            .unwrap();
+
+        assert_eq!(state.cycles_written, 3);
+        assert_eq!(state.mic.samples.len(), 3 * frame);
+        assert_eq!(state.mix.as_ref().unwrap().samples.len(), 3 * frame);
+        assert_eq!(
+            &state.mix.as_ref().unwrap().samples[2 * frame..],
+            &[0, 0, 0]
+        );
+        assert_eq!(state.peers.get(&early).unwrap().samples.len(), 3 * frame);
+        assert_eq!(
+            &state.peers.get(&early).unwrap().samples[2 * frame..],
+            &[2, 2, 2]
+        );
+        assert_eq!(
+            state.peers.get(&late).unwrap().samples,
+            vec![0, 0, 0, 0, 0, 0, 7, 7, 7],
+            "late peer is back-padded by completed cycles before this batch"
+        );
+    }
+
+    #[test]
+    fn skipped_batches_keep_all_tracks_equal_length() {
+        let frame = 2;
+        let p1 = an_id();
+        let p2 = an_id();
+        let mut state = test_writer_state(frame, true);
+
+        let first = test_batch(
+            vec![
+                NewPeer {
+                    id: p1,
+                    filename: "p1.wav".to_string(),
+                },
+                NewPeer {
+                    id: p2,
+                    filename: "p2.wav".to_string(),
+                },
+            ],
+            vec![1; frame],
+            Some(vec![5; frame]),
+            vec![(p1, vec![10; frame]), (p2, vec![20; frame])],
+        );
+        state
+            .apply_batch(&first, |_| Ok(TestWriter::default()))
+            .unwrap();
+
+        let _dropped_cycle = test_batch(
+            Vec::new(),
+            vec![2; frame],
+            Some(vec![6; frame]),
+            vec![(p1, vec![11; frame])],
+        );
+
+        let after_drop = test_batch(
+            Vec::new(),
+            vec![3; frame],
+            None,
+            vec![(p1, vec![12; frame])],
+        );
+        state
+            .apply_batch(&after_drop, |_| Ok(TestWriter::default()))
+            .unwrap();
+
+        let expected = 2 * frame;
+        assert_eq!(state.cycles_written, 2);
+        assert_eq!(state.mic.samples.len(), expected);
+        assert_eq!(state.mix.as_ref().unwrap().samples.len(), expected);
+        assert_eq!(state.peers.get(&p1).unwrap().samples.len(), expected);
+        assert_eq!(state.peers.get(&p2).unwrap().samples.len(), expected);
+        assert_eq!(
+            &state.peers.get(&p2).unwrap().samples[frame..],
+            &[0, 0],
+            "peer absent from an applied batch gets silence for that cycle"
+        );
+    }
+
     #[test]
     fn all_tracks_equal_length_after_n_cycles() {
         let dir = tmpdir("equal");
@@ -406,4 +731,23 @@ mod tests {
             "no mix track in stems-only mode"
         );
     }
+
+    #[cfg(unix)]
+    #[test]
+    fn async_peer_create_error_surfaces_at_finalize() {
+        use std::os::unix::fs::PermissionsExt;
+
+        let dir = tmpdir("asyncerr");
+        let mut rec = MultitrackRecorder::create(&dir, 4, false).unwrap();
+        std::fs::set_permissions(&dir, std::fs::Permissions::from_mode(0o500)).unwrap();
+
+        rec.add_peer(an_id(), "blocked").unwrap();
+        rec.end_cycle().unwrap();
+        let result = rec.finalize();
+
+        std::fs::set_permissions(&dir, std::fs::Permissions::from_mode(0o700)).unwrap();
+        let err = result.unwrap_err();
+        assert_eq!(err.kind(), io::ErrorKind::PermissionDenied);
+        let _ = std::fs::remove_dir_all(&dir);
+    }
 }

src/codec/mod.rs

@@ -20,6 +20,9 @@ pub trait AudioDecoder: Send {
     /// If `compressed` is `None` (or `Some(&[])`), it indicates packet loss,
     /// enabling the decoder to perform packet loss concealment (PLC).
     fn decode(&mut self, compressed: Option<&[u8]>) -> Result<Vec<i16>, CodecError>;
+
+    /// Reconstructs the previous lost frame from the next packet's in-band FEC.
+    fn decode_fec(&mut self, next_payload: &[u8]) -> Result<Vec<i16>, CodecError>;
 }
 
 pub mod opus_impl;

src/codec/opus_impl.rs

@@ -1,5 +1,49 @@
 use crate::codec::{AudioDecoder, AudioEncoder, CodecError};
-use opus::{Application, Channels, Decoder, Encoder};
+use crate::config::AudioProfile;
+use opus::{Application, Bitrate, Channels, Decoder, Encoder};
+
+/// Concrete libopus encoder settings derived from an [`AudioProfile`]. Plain
+/// data, so the profile→params mapping ([`opus_params`]) stays a pure,
+/// unit-testable function (W12).
+#[derive(Debug, Clone, Copy, PartialEq, Eq)]
+pub struct OpusParams {
+    /// Target bitrate in bits/sec.
+    pub bitrate: i32,
+    /// Enable in-band forward error correction (loss redundancy in the bitstream).
+    pub inband_fec: bool,
+    /// Expected packet-loss percentage (0..=100); tunes how much FEC libopus adds.
+    pub packet_loss_perc: i32,
+    /// Discontinuous transmission: stop sending during silence to save bandwidth.
+    pub dtx: bool,
+}
+
+/// Map a named profile to concrete Opus parameters. Pure — the W12 testable seam.
+///
+/// `BadNetwork` deliberately runs a *lower* bitrate than `Balanced`: in-band FEC
+/// redundancy is carried inside the same bitstream, so trimming the base bitrate
+/// leaves headroom for the redundancy on a congested link.
+pub fn opus_params(profile: AudioProfile) -> OpusParams {
+    match profile {
+        AudioProfile::LowLatency => OpusParams {
+            bitrate: 24_000,
+            inband_fec: false,
+            packet_loss_perc: 0,
+            dtx: false,
+        },
+        AudioProfile::Balanced => OpusParams {
+            bitrate: 32_000,
+            inband_fec: true,
+            packet_loss_perc: 10,
+            dtx: false,
+        },
+        AudioProfile::BadNetwork => OpusParams {
+            bitrate: 20_000,
+            inband_fec: true,
+            packet_loss_perc: 25,
+            dtx: false,
+        },
+    }
+}
 
 pub struct OpusEncoder {
     encoder: Encoder,
@@ -17,6 +61,29 @@ impl OpusEncoder {
             .map_err(|e| CodecError::Init(format!("Failed to create Opus encoder: {}", e)))?;
         Ok(Self { encoder })
     }
+
+    /// Apply concrete codec parameters to the live encoder. Safe to call between
+    /// frames, so the user can switch profile mid-call.
+    pub fn apply_params(&mut self, params: &OpusParams) -> Result<(), CodecError> {
+        self.encoder
+            .set_bitrate(Bitrate::Bits(params.bitrate))
+            .map_err(|e| CodecError::Init(format!("set_bitrate: {}", e)))?;
+        self.encoder
+            .set_inband_fec(params.inband_fec)
+            .map_err(|e| CodecError::Init(format!("set_inband_fec: {}", e)))?;
+        self.encoder
+            .set_packet_loss_perc(params.packet_loss_perc)
+            .map_err(|e| CodecError::Init(format!("set_packet_loss_perc: {}", e)))?;
+        self.encoder
+            .set_dtx(params.dtx)
+            .map_err(|e| CodecError::Init(format!("set_dtx: {}", e)))?;
+        Ok(())
+    }
+
+    /// Apply a named [`AudioProfile`] (shorthand for `apply_params(&opus_params(p))`).
+    pub fn apply_profile(&mut self, profile: AudioProfile) -> Result<(), CodecError> {
+        self.apply_params(&opus_params(profile))
+    }
 }
 
 impl AudioEncoder for OpusEncoder {
@@ -95,12 +162,64 @@ impl AudioDecoder for OpusDecoder {
         pcm.truncate(decoded_per_channel * channels_count);
         Ok(pcm)
     }
+
+    fn decode_fec(&mut self, next_payload: &[u8]) -> Result<Vec<i16>, CodecError> {
+        let channels_count = self.channels_count();
+        let mut pcm = vec![0i16; self.frame_samples * channels_count];
+
+        let decoded_per_channel = self
+            .decoder
+            .decode(next_payload, &mut pcm, true)
+            .map_err(|e| CodecError::Decode(format!("Opus FEC decoding failed: {}", e)))?;
+
+        pcm.truncate(decoded_per_channel * channels_count);
+        Ok(pcm)
+    }
 }
 
 #[cfg(test)]
 mod tests {
     use super::*;
 
+    #[test]
+    fn test_opus_params_mapping() {
+        let low = opus_params(AudioProfile::LowLatency);
+        let bal = opus_params(AudioProfile::Balanced);
+        let bad = opus_params(AudioProfile::BadNetwork);
+
+        // LowLatency has no loss redundancy; the other two do.
+        assert!(!low.inband_fec);
+        assert_eq!(low.packet_loss_perc, 0);
+        assert!(bal.inband_fec);
+        assert!(bad.inband_fec);
+
+        // Capture-side gating suppresses silence; no profile adds Opus DTX.
+        assert!(!low.dtx && !bal.dtx && !bad.dtx);
+        assert!(bad.packet_loss_perc > bal.packet_loss_perc);
+
+        // BadNetwork trims base bitrate to make room for FEC redundancy.
+        assert!(bad.bitrate < bal.bitrate);
+
+        // All bitrates are sane positive voice rates.
+        for p in [low, bal, bad] {
+            assert!(p.bitrate > 0 && p.bitrate <= 64_000);
+            assert!((0..=100).contains(&p.packet_loss_perc));
+        }
+    }
+
+    #[test]
+    fn test_apply_profile_sets_bitrate() {
+        let mut encoder = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
+        // Every profile applies cleanly to a real encoder...
+        for profile in AudioProfile::ALL {
+            encoder.apply_profile(profile).unwrap();
+        }
+        // ...and the last-applied bitrate is reflected by the encoder.
+        encoder.apply_profile(AudioProfile::Balanced).unwrap();
+        let want = opus_params(AudioProfile::Balanced).bitrate;
+        assert_eq!(encoder.encoder.get_bitrate().unwrap(), Bitrate::Bits(want));
+    }
+
     #[test]
     fn test_round_trip() {
         let mut encoder = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();

src/config.rs

@@ -93,6 +93,60 @@ impl std::fmt::Display for RecordingMode {
     }
 }
 
+/// Named Opus encoder / network-resilience policy (W12). The user picks a
+/// profile instead of raw codec knobs; the concrete libopus parameters live in
+/// `codec::opus_impl::opus_params`. Applies live to the running encoder.
+#[derive(Debug, Clone, Copy, PartialEq, Eq, Serialize, Deserialize, Default)]
+pub enum AudioProfile {
+    /// Lowest mouth-to-ear delay: modest bitrate, no FEC redundancy. Best on a
+    /// clean LAN / low-loss link where added latency matters more than loss.
+    LowLatency,
+    /// Sensible default: voice bitrate with in-band FEC for light packet loss.
+    #[default]
+    Balanced,
+    /// Maximum resilience on a lossy/congested link: in-band FEC tuned for heavy
+    /// loss, at a lower bitrate to leave headroom for the redundancy.
+    BadNetwork,
+}
+
+impl AudioProfile {
+    /// All variants, in picker display order.
+    pub const ALL: [AudioProfile; 3] = [
+        AudioProfile::LowLatency,
+        AudioProfile::Balanced,
+        AudioProfile::BadNetwork,
+    ];
+
+    /// Compact discriminant for handing the profile to the capture thread via an
+    /// atomic. Pairs with [`AudioProfile::from_u8`].
+    pub fn as_u8(self) -> u8 {
+        match self {
+            AudioProfile::LowLatency => 0,
+            AudioProfile::Balanced => 1,
+            AudioProfile::BadNetwork => 2,
+        }
+    }
+
+    /// Inverse of [`AudioProfile::as_u8`]; unknown values fall back to the default.
+    pub fn from_u8(v: u8) -> AudioProfile {
+        match v {
+            0 => AudioProfile::LowLatency,
+            2 => AudioProfile::BadNetwork,
+            _ => AudioProfile::Balanced,
+        }
+    }
+}
+
+impl std::fmt::Display for AudioProfile {
+    fn fmt(&self, f: &mut std::fmt::Formatter<'_>) -> std::fmt::Result {
+        f.write_str(match self {
+            AudioProfile::LowLatency => "Low latency",
+            AudioProfile::Balanced => "Balanced",
+            AudioProfile::BadNetwork => "Bad network",
+        })
+    }
+}
+
 impl std::fmt::Display for RoomLayout {
     fn fmt(&self, f: &mut std::fmt::Formatter<'_>) -> std::fmt::Result {
         f.write_str(match self {
@@ -199,6 +253,10 @@ pub struct AppConfig {
     pub clip_volume_universal: bool,
     #[serde(default)]
     pub network_mode: NetworkMode,
+    /// Opus encoder / network-resilience profile (W12). Applies live to the
+    /// running encoder; default `Balanced`.
+    #[serde(default)]
+    pub audio_profile: AudioProfile,
     /// Presence posture for the friends idle listener (W7): invisible / normal /
     /// discoverable. Default `Normal` = answer friends only, no DNS beacon.
     #[serde(default)]
@@ -380,6 +438,7 @@ impl Default for AppConfig {
             show_player_bar: true,
             clip_volume_universal: true,
             network_mode: NetworkMode::default(),
+            audio_profile: AudioProfile::default(),
             presence_mode: crate::presence::PresenceMode::default(),
             echo_cancellation_enabled: false,
             notifications_enabled: true,
@@ -1019,6 +1078,38 @@ mod tests {
         assert_ne!(display_0, display_2);
     }
 
+    #[test]
+    fn test_audio_profile() {
+        // Default is Balanced.
+        assert_eq!(AudioProfile::default(), AudioProfile::Balanced);
+
+        // ALL holds the three variants.
+        assert_eq!(AudioProfile::ALL.len(), 3);
+        assert!(AudioProfile::ALL.contains(&AudioProfile::LowLatency));
+        assert!(AudioProfile::ALL.contains(&AudioProfile::Balanced));
+        assert!(AudioProfile::ALL.contains(&AudioProfile::BadNetwork));
+
+        // as_u8 / from_u8 round-trip every variant, and unknown bytes fall back
+        // to the default rather than panicking.
+        for p in AudioProfile::ALL {
+            assert_eq!(AudioProfile::from_u8(p.as_u8()), p);
+        }
+        assert_eq!(AudioProfile::from_u8(99), AudioProfile::Balanced);
+
+        // serde round-trips, and Display strings are non-empty + distinct.
+        let mut labels = Vec::new();
+        for p in AudioProfile::ALL {
+            let s = serde_json::to_string(&p).unwrap();
+            assert_eq!(serde_json::from_str::<AudioProfile>(&s).unwrap(), p);
+            let label = p.to_string();
+            assert!(!label.is_empty());
+            labels.push(label);
+        }
+        labels.sort();
+        labels.dedup();
+        assert_eq!(labels.len(), 3);
+    }
+
     #[test]
     fn test_unknown_field_tolerance() {
         // Unknown/extra field tolerance: a config JSON containing an extra unrecognized key should still deserialize.

src/core/jitter.rs

@@ -202,11 +202,15 @@ impl JitterBuffer {
                     None
                 } else {
                     // Gap with later packets already buffered: a packet was lost
-                    // or reordered out of window. Conceal this frame via Opus PLC
+                    // or reordered out of window. First try Opus in-band FEC from
-                    // and grow the cushion — the jitter beat our current delay.
+                    // the next packet; if unavailable, fall back to plain PLC.
                     self.next_seq = Some(next.wrapping_add(1));
                     self.note_disruption();
-                    self.decoder.decode(None).ok()
+                    let next_payload = self.packets.values().next().expect("non-empty");
+                    self.decoder
+                        .decode_fec(next_payload)
+                        .or_else(|_| self.decoder.decode(None))
+                        .ok()
                 }
             }
         }
@@ -221,8 +225,8 @@ impl JitterBuffer {
 #[cfg(test)]
 mod tests {
     use super::*;
-    use crate::codec::AudioEncoder;
+    use crate::codec::opus_impl::{OpusDecoder, OpusEncoder, OpusParams};
-    use crate::codec::opus_impl::OpusEncoder;
+    use crate::codec::{AudioDecoder, AudioEncoder};
     use opus::{Application, Channels};
 
     /// A real, decodable Opus packet for one 20ms mono frame at amplitude `amp`.
@@ -233,6 +237,32 @@ mod tests {
         enc.encode(&pcm).unwrap()
     }
 
+    fn tone_frame(enc: &mut OpusEncoder, amp: i16, frame_index: usize) -> Vec<u8> {
+        let pcm: Vec<i16> = (0..FRAME_SAMPLES)
+            .map(|i| {
+                let sample_index = frame_index * FRAME_SAMPLES + i;
+                let t = sample_index as f32 / 48_000.0;
+                let fundamental = (t * 220.0 * 2.0 * std::f32::consts::PI).sin();
+                let harmonic = (t * 440.0 * 2.0 * std::f32::consts::PI).sin();
+                ((fundamental * 0.7 + harmonic * 0.3) * amp as f32) as i16
+            })
+            .collect();
+        enc.encode(&pcm).unwrap()
+    }
+
+    fn rms_error(a: &[i16], b: &[i16]) -> f64 {
+        assert_eq!(a.len(), b.len());
+        let sum_sq: f64 = a
+            .iter()
+            .zip(b)
+            .map(|(&left, &right)| {
+                let diff = left as f64 - right as f64;
+                diff * diff
+            })
+            .sum();
+        (sum_sq / a.len() as f64).sqrt()
+    }
+
     #[test]
     fn buffers_then_plays_in_order() {
         let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
@@ -289,6 +319,66 @@ mod tests {
         assert!(jb.pop_frame().is_none());
     }
 
+    #[test]
+    fn uses_in_band_fec_from_next_packet_for_gap() {
+        let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
+        enc.apply_params(&OpusParams {
+            bitrate: 20_000,
+            inband_fec: true,
+            packet_loss_perc: 60,
+            dtx: false,
+        })
+        .unwrap();
+
+        let dropped_seq = 5usize;
+        let amps = [1800, 1800, 1800, 1800, 1800, 12_000, 12_000, 12_000];
+        let packets: Vec<Vec<u8>> = amps
+            .into_iter()
+            .enumerate()
+            .map(|(seq, amp)| tone_frame(&mut enc, amp, seq))
+            .collect();
+
+        let mut expected_decoder = OpusDecoder::new(48000, Channels::Mono, FRAME_SAMPLES).unwrap();
+        for packet in packets.iter().take(dropped_seq) {
+            expected_decoder.decode(Some(packet)).unwrap();
+        }
+        let expected_lost = expected_decoder
+            .decode(Some(&packets[dropped_seq]))
+            .unwrap();
+
+        let mut plc_decoder = OpusDecoder::new(48000, Channels::Mono, FRAME_SAMPLES).unwrap();
+        for packet in packets.iter().take(dropped_seq) {
+            plc_decoder.decode(Some(packet)).unwrap();
+        }
+        let pure_plc = plc_decoder.decode(None).unwrap();
+
+        let mut jb = JitterBuffer::new().unwrap();
+        for (seq, packet) in packets.iter().enumerate() {
+            if seq != dropped_seq {
+                jb.insert(seq as u32, packet.clone());
+            }
+        }
+
+        for _ in 0..dropped_seq {
+            assert_eq!(jb.pop_frame().map(|frame| frame.len()), Some(FRAME_SAMPLES));
+        }
+
+        let recovered = jb.pop_frame().expect("gap should be reconstructed");
+        assert_eq!(recovered.len(), FRAME_SAMPLES);
+        assert!(
+            jb.packets.contains_key(&(dropped_seq as u32 + 1)),
+            "FEC source packet must remain buffered for normal decode"
+        );
+        assert_eq!(jb.pop_frame().map(|frame| frame.len()), Some(FRAME_SAMPLES));
+
+        let fec_error = rms_error(&recovered, &expected_lost);
+        let plc_error = rms_error(&pure_plc, &expected_lost);
+        assert!(
+            fec_error < plc_error * 0.75,
+            "FEC reconstruction should be materially closer than PLC (fec_error={fec_error}, plc_error={plc_error})"
+        );
+    }
+
     #[test]
     fn drops_packets_already_played() {
         let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();

src/core/messages.rs

@@ -1,4 +1,4 @@
-use crate::config::{NetworkMode, RecordingMode};
+use crate::config::{AudioProfile, NetworkMode, RecordingMode};
 use crate::friends::Friend;
 use crate::network::PeerState;
 use crate::presence::{FriendPresence, PresenceMode};
@@ -56,6 +56,10 @@ pub enum CoreCommand {
     /// Set the relay/discovery posture. Takes effect on the next room join,
     /// since the endpoint is (re)built then.
     SetNetworkMode(NetworkMode),
+    /// Set the Opus encoder / network-resilience profile (W12). Applies live to
+    /// the running capture encoder, and to the next call's encoder. Sent at
+    /// startup from config and whenever the user changes it.
+    SetAudioProfile(AudioProfile),
     /// Start/stop recording the call to a local WAV (your mic + the incoming
     /// mix). No-op start if already recording / not in a call.
     SetRecording(bool),
@@ -212,6 +216,7 @@ pub fn delivery_class(cmd: &CoreCommand) -> DeliveryClass {
             input_device: _,
         }
         | CoreCommand::SetNetworkMode(_)
+        | CoreCommand::SetAudioProfile(_)
         | CoreCommand::SetRecording(_)
         | CoreCommand::SetRecordingMode(_)
         | CoreCommand::SendChat(_)
@@ -292,6 +297,7 @@ pub fn coalesce_key(cmd: &CoreCommand) -> Option<CoalesceKey> {
             input_device: _,
         }
         | CoreCommand::SetNetworkMode(_)
+        | CoreCommand::SetAudioProfile(_)
         | CoreCommand::SetRecording(_)
         | CoreCommand::SetRecordingMode(_)
         | CoreCommand::SendChat(_)
@@ -574,6 +580,7 @@ mod tests {
             },
             CoreCommand::SetPeerMuted(peer, true),
             CoreCommand::SetPresenceMode(PresenceMode::Normal),
+            CoreCommand::SetAudioProfile(crate::config::AudioProfile::BadNetwork),
             CoreCommand::SendChat("hello".to_string()),
         ];
 

src/core/mod.rs

@@ -17,7 +17,7 @@ use crate::network::{
 };
 
 use crate::audio::multitrack::MultitrackRecorder;
-use crate::config::{NetworkMode, RecordingMode};
+use crate::config::{AudioProfile, NetworkMode, RecordingMode};
 use crate::presence::PresenceMode;
 use iroh::{
     Endpoint, EndpointAddr, EndpointId, RelayMode, SecretKey, endpoint::presets, protocol::Router,
@@ -31,6 +31,12 @@ use tokio::sync::{Mutex, mpsc};
 
 type CoalesceStore = Arc<StdMutex<HashMap<CoalesceKey, CoreCommand>>>;
 
+// Mixer -> playback-worker handoff. The playback ring itself targets three
+// 20ms frames; allow at most two more in flight so worker lag applies
+// backpressure before the ring can overshoot to its 200ms cap (A6).
+const PLAYBACK_HANDOFF_QUEUE_FRAMES: usize = 2;
+const PLAYBACK_HANDOFF_RETRY: Duration = Duration::from_millis(1);
+
 pub struct CoreController {
     reliable_tx: mpsc::UnboundedSender<CoreCommand>,
     coalesce: CoalesceStore,
@@ -156,6 +162,22 @@ fn audio_datagram_len_ok(len: usize) -> bool {
     (4..=4 + MAX_OPUS_PAYLOAD).contains(&len)
 }
 
+async fn send_playback_frame(
+    tx: &std::sync::mpsc::SyncSender<Vec<i16>>,
+    mut frame: Vec<i16>,
+) -> bool {
+    loop {
+        match tx.try_send(frame) {
+            Ok(()) => return true,
+            Err(std::sync::mpsc::TrySendError::Full(returned)) => {
+                frame = returned;
+                tokio::time::sleep(PLAYBACK_HANDOFF_RETRY).await;
+            }
+            Err(std::sync::mpsc::TrySendError::Disconnected(_)) => return false,
+        }
+    }
+}
+
 /// The presence label to broadcast for a detected game: its display name,
 /// sanitized + length-capped, or `None` when there's no game or no broadcastable
 /// name (a Steam appid without a manifest name, or a label that sanitizes empty).
@@ -1162,6 +1184,12 @@ async fn run_core_loop(
     // App-internal capture/playback gains (f32 bits), live-read by the audio loops.
     let input_gain = Arc::new(std::sync::atomic::AtomicU32::new(1.0f32.to_bits()));
     let output_gain = Arc::new(std::sync::atomic::AtomicU32::new(1.0f32.to_bits()));
+    // Opus encoder profile (W12) as a discriminant, live-read by the capture
+    // thread so a mid-call profile switch re-tunes the running encoder. Set from
+    // config via the GUI's startup `SetAudioProfile`; defaults to Balanced.
+    let audio_profile = Arc::new(std::sync::atomic::AtomicU8::new(
+        AudioProfile::default().as_u8(),
+    ));
     // Call recording: an optional live recorder (mic FIFO + WAV writer), shared
     // by the capture thread (pushes mic) and the mixer task (writes mix frames).
     // `is_recording` is a fast-path gate so the audio loops only take the lock
@@ -1648,7 +1676,8 @@ async fn run_core_loop(
 
                 // Setup raw audio channels
                 let (capture_tx, capture_rx) = std::sync::mpsc::channel();
-                let (playback_tx, playback_rx) = std::sync::mpsc::channel();
+                let (playback_tx, playback_rx) =
+                    std::sync::mpsc::sync_channel(PLAYBACK_HANDOFF_QUEUE_FRAMES);
 
                 // Echo cancellation: if enabled, load PipeWire's echo-cancel module
                 // bound to the chosen real devices and route capture/playback
@@ -1740,6 +1769,7 @@ async fn run_core_loop(
                 let is_recording_capture = is_recording.clone();
                 let multitrack_capture = multitrack.clone();
                 let is_multitrack_capture = is_multitrack.clone();
+                let audio_profile_capture = audio_profile.clone();
 
                 let capture_thread = std::thread::spawn(move || {
                     use opus::{Application, Channels};
@@ -1751,6 +1781,13 @@ async fn run_core_loop(
                                 return;
                             }
                         };
+                    // Tune the encoder to the configured profile (W12), then track
+                    // the live discriminant so a mid-call switch re-applies it.
+                    let mut current_profile =
+                        AudioProfile::from_u8(audio_profile_capture.load(Ordering::Relaxed));
+                    if let Err(e) = encoder.apply_profile(current_profile) {
+                        crate::log_msg(&format!("Opus profile apply failed: {:?}", e));
+                    }
                     // Per-sender packet sequence number, prepended to every frame so
                     // receivers can reorder and conceal loss. Wraps after ~years.
                     let mut seq: u32 = 0;
@@ -1763,6 +1800,14 @@ async fn run_core_loop(
                     let mut mic_meter = MicLevelMeter::new();
 
                     while let Ok(mut pcm) = capture_rx.recv() {
+                        // Re-tune the encoder if the user switched profile mid-call.
+                        // Cheap atomic load per frame; only reconfigures on change.
+                        let want =
+                            AudioProfile::from_u8(audio_profile_capture.load(Ordering::Relaxed));
+                        if want != current_profile && encoder.apply_profile(want).is_ok() {
+                            current_profile = want;
+                        }
+
                         // Apply the input gain first so the meter, gate, and what we
                         // transmit all reflect the same (gained) signal.
                         apply_volume(
@@ -2092,7 +2137,7 @@ async fn run_core_loop(
                             mixed
                         };
 
-                        if playback_tx.send(frame_to_send).is_err() {
+                        if !send_playback_frame(&playback_tx, frame_to_send).await {
                             break;
                         }
 
@@ -2682,6 +2727,13 @@ async fn run_core_loop(
                 }
             }
 
+            CoreCommand::SetAudioProfile(profile) => {
+                // Publish the new profile to the capture thread (W12). It picks up
+                // the change on its next frame and re-tunes the live encoder; a
+                // call that starts later reads the same atomic at encoder creation.
+                audio_profile.store(profile.as_u8(), Ordering::Relaxed);
+            }
+
             CoreCommand::RegenerateIdentity => {
                 // Mint + persist a fresh identity, discarding the old one. The
                 // persistent endpoint is rebuilt with the new key (now if idle, else
@@ -3193,12 +3245,15 @@ async fn run_core_loop(
 mod tests {
     use super::{
         KnownPeers, MAX_OPUS_PAYLOAD, MAX_RETAINED_PEERS, MIC_LEVEL_REPORT_SAMPLES, MicLevelMeter,
-        PeerSpeakTicket, admit_retained, apply_peer_volume, apply_volume, audio_datagram_len_ok,
+        PLAYBACK_HANDOFF_QUEUE_FRAMES, PeerSpeakTicket, admit_retained, apply_peer_volume,
-        coalesce_insert, coalesce_pop, frame_level, mix_frames, mix_stereo_frames,
+        apply_volume, audio_datagram_len_ok, coalesce_insert, coalesce_pop, frame_level,
-        next_game_change, should_auto_fetch, stereo_to_mono,
+        mix_frames, mix_stereo_frames, next_game_change, send_playback_frame, should_auto_fetch,
+        stereo_to_mono,
     };
     use crate::core::messages::{CoalesceKey, CoreCommand, coalesce_key};
     use std::collections::{HashMap, HashSet};
+    use std::sync::mpsc::sync_channel;
+    use std::time::Duration;
 
     fn endpoint_id() -> iroh::EndpointId {
         iroh::SecretKey::generate().public()
@@ -3432,6 +3487,33 @@ mod tests {
         assert!(!audio_datagram_len_ok(5 + MAX_OPUS_PAYLOAD));
     }
 
+    #[tokio::test]
+    async fn playback_handoff_waits_for_bounded_queue_space() {
+        let (tx, rx) = sync_channel::<Vec<i16>>(PLAYBACK_HANDOFF_QUEUE_FRAMES);
+        for n in 0..PLAYBACK_HANDOFF_QUEUE_FRAMES {
+            tx.try_send(vec![n as i16]).unwrap();
+        }
+
+        let worker = std::thread::spawn(move || {
+            std::thread::sleep(Duration::from_millis(20));
+            for n in 0..PLAYBACK_HANDOFF_QUEUE_FRAMES {
+                assert_eq!(rx.recv().unwrap(), vec![n as i16]);
+            }
+            assert_eq!(rx.recv().unwrap(), vec![99, 100]);
+        });
+
+        let sent = tokio::time::timeout(
+            Duration::from_secs(1),
+            send_playback_frame(&tx, vec![99, 100]),
+        )
+        .await
+        .expect("bounded handoff should unblock after the worker drains a frame");
+
+        assert!(sent);
+        drop(tx);
+        worker.join().unwrap();
+    }
+
     #[test]
     fn mic_meter_holds_the_peak_across_the_window() {
         let mut m = MicLevelMeter::new();