The repo never enforced rustfmt, so formatting had drifted broadly. This is a
single mechanical `cargo fmt` pass over the whole crate (no behavioral change;
lib suite green, 493 passed). Going forward fmt should be enforced (planned CI
fmt --check step). Part of the 0.6.1 hygiene pass.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
- src/audio/resample.rs: pure linear PushResampler (capture) +
StereoPullResampler (playback pull), 6 unit tests green on Linux.
- choose_config: prefer native 48kHz, else fall back to device default
config and convert at the boundary instead of hard-erroring.
- run_capture: resample device-rate mono -> 48kHz on the drain thread.
- i16<->f32 helpers. Playback build_output remap still TODO (Codex).
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Three Windows-compatibility fixes from the Codex review. Implemented by Codex
(gpt-5.5); reviewed and committed by Claude.
W5 — echo cancellation is a Linux/PipeWire feature, but the toggle was shown and
live on Windows, so a Windows join tried `pactl` and errored before falling back.
Now `#[cfg(target_os = "linux")]` gates the core enable path (and the
ActiveSession guard field); on other targets the Settings + in-call controls
render as a disabled checkbox with a "not available on Windows yet" note.
W6 — pixelpass PATH lookup only tried `pixelpass`; on Windows it now also tries
`pixelpass.exe` via a cfg-selected candidate list (+ unit test).
W9 — the Linux audio stack (pipewire/pw_cli/echo_cancel/audio_probe + the
`PlatformAudioBackend` alias and device-enum re-export) was gated `cfg(unix)`;
tightened to `cfg(target_os = "linux")` so a hypothetical macOS build won't try
to compile PipeWire. cpal stays `cfg(windows)`. Genuinely-Unix file/key
permission code in lib.rs/identity.rs left as `cfg(unix)`.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Give the Windows device pickers a real device list (Phase 0/1 left pw_cli
returning nothing off-Linux) and generalize enumeration into a
platform-neutral interface.
- audio/mod.rs: move AudioDevice here (neutral home), gate pw_cli to
cfg(unix), and re-export enumerate_audio_devices per-platform (pw_cli on
unix, cpal_impl on windows). Also drop a now-stale "no-op stub" doc note.
- cpal_impl.rs: add enumerate_audio_devices() — iterate the cpal host's
input + output devices into AudioDevice (name == description == the cpal
friendly name, which is what resolve() matches target_node against, so a
saved selection round-trips), sorted by description.
- pw_cli.rs: use super::AudioDevice instead of a local copy; parsing +
tests unchanged.
- app/mod.rs: one-line import change; the device-picker logic is untouched.
Verified: shipped Linux state green (build --locked, clippy, 316/316,
pw_cli parse tests 6/6); the cpal enumerator compiles against real cpal via
the Linux/ALSA toggle. Runtime device listing on Windows is pending a real
host (M2/M3). WASAPI names are less stable than PipeWire node names, so a
saved device may not always round-trip (falls back to default).
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Make the tree compile for Windows without touching core logic, by
confining all Linux/PipeWire assumptions behind cfg gates and a single
platform-selected backend alias. No new dependencies — the cpal/WASAPI
backend lands in Phase 1; this ships a no-op stub.
- Cargo.toml: move pipewire + rfd(xdg-portal) under cfg(unix); add a
cfg(windows) rfd using the Win32 dialog backend.
- audio: gate pipewire_impl to unix, add a cpal_impl stub for windows,
and select between them via the new PlatformAudioBackend alias.
- core: use PlatformAudioBackend instead of the concrete PipeWireBackend.
- lib: gate the unix-only 0o600 log-file mode code (+ its test); Windows
logs inherit the directory ACL.
- audio_probe: gate this PipeWire diagnostic to unix with a stub main.
- app: open URLs via rundll32 on windows, xdg-open on unix (shell-free).
- ci: add .gitea/workflows/windows-build.yml (M1) — build + lib tests for
x86_64-pc-windows-msvc, with CMAKE_POLICY_VERSION_MINIMUM=3.5 for the
vendored libopus build. Needs a windows act_runner to actually run.
Linux build/clippy/tests green (316/316). The Windows path is verified by
inspection only (no local Windows toolchain); CI is the real gate.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Scopes the differentiating "record every peer to their own synced track"
feature and lands its pure, isolated core (no live-audio wiring yet).
- docs/multitrack-recording-plan.md: scope contract + locked decisions
(raw stems pre-volume/mute, stems + a mixed track, silence-pad late joiners).
- src/audio/multitrack.rs: MultitrackRecorder over the existing WavWriter.
One master clock = the mixer cycle; every end_cycle() appends exactly
FRAME_SAMPLES to every track (silence where idle) so all stems stay
sample-aligned. add_peer back-pads a late joiner to cycle 0; track_filename
gives fs-safe `<slug>-<shortid>.wav` (reuses sanitize_name). Optional mix
track for "Both" mode.
- +5 unit tests: equal length across tracks, late-joiner leading silence,
stems-only omits mix, fit() pad/truncate, filename slugging/disambiguation.
Stage 2 (wire into the mixer) is next, behind a checkpoint. 168 lib tests,
clippy --all-targets clean.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Opt-in recording of the full call as you experienced it. New dep-free
src/audio/recorder.rs: a canonical mono S16LE WavWriter (header patched on
finalize) plus a Recorder that buffers your transmitted mic in a bounded FIFO
and sums it, sample-aligned, with each incoming-mix frame the playout mixer
produces. The two independently-clocked streams stay aligned via the FIFO
(capped at ~200ms so drift lag can't grow without bound); silent stretches
record the incoming mix alone. Dep-free UTC timestamp -> sortable filename.
Wiring: CoreCommand::SetRecording toggles an Arc<Mutex<Option<Recorder>>> gated
by an is_recording flag (so the capture/mixer hot paths only lock while actually
recording); capture pushes post-gate mic, the mixer writes the pre-deafen mix.
Recording finalizes on stop, room leave, and room switch. UI: a Record/Stop
button in the controls and a red "● REC m:ss" pill in the room header;
core-confirmed Recording{Started,Stopped} events drive the UI flag so a failed
start can't lie. Files land in ~/peerspeak-recordings/.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Replace the mixer's per-sample hard clamp with a lossless i32 bus sum fed
through a feed-forward soft limiter (instant attack, ~120ms release). Below
the ceiling it's transparent and sample-exact; loud multi-peer moments are
ridden down to the ceiling instead of shattering into hard-clip distortion.
State carries across frames so a sustained-loud stretch doesn't re-attack
every 20ms frame. The master output gain now applies inside the limiter so a
boost past the ceiling is limited too.
mix_frames now returns the lossless i32 sum (saturation responsibility moved
to the limiter); its tests assert losslessness, and the new limiter module
carries the saturation/transparency/release guarantees.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Add an opt-in Echo Cancellation toggle (Settings) that routes the call
through PipeWire's module-echo-cancel (WebRTC AEC + noise suppression + AGC)
instead of running an in-process canceller. PipeWire already sees both the
mic and the speaker monitor, so it handles the echo-reference alignment for
free and we avoid a C++ DSP dependency.
src/audio/echo_cancel.rs (new):
- enable(real_source, real_sink) loads the module via pactl (aec_method=webrtc),
bound to the chosen devices with source_master/sink_master (defaults if unset),
waits for the virtual nodes to appear, and returns an RAII guard that unloads
the module on drop. Best-effort pre-clean of a stale instance from a crashed run.
- EC_SOURCE / EC_SINK are the virtual cleaned-mic source and reference sink.
core: when echo_cancellation is set on Join, load the module and point capture
at EC_SOURCE / playback at EC_SINK; stash the guard in ActiveSession so it
unloads on shutdown (after the audio streams release the nodes). Any failure
logs + warns the UI and falls back to the direct devices — never blocks the call.
config: new echo_cancellation_enabled (serde default false). app: Settings
checkbox under Mic Sensitivity, applied on next room join. An ignored live
smoke test covers the real load/unload path.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Replace the per-frame hard-cut noise gate with a stateful envelope gate
(src/audio/gate.rs):
- Hysteresis: opens at the slider threshold, closes only below 0.6x that,
so speech near the threshold doesn't flap the gate.
- Attack/release: per-sample gain ramp (5ms open, 80ms close) instead of a
click — the gate fades rather than dropping frames outright.
- Hangover: holds the gate open 200ms after the level drops, so word tails
and brief pauses aren't chopped.
A fully-closed frame still reports don't-transmit, preserving the original
bandwidth win of not sending pure silence (receiver jitter buffer conceals
the gap). The live slider value is read per frame so changes apply
immediately. Settings slider gains a one-line hint. 6 unit tests cover
attack shape, hysteresis, hangover-then-release, disabled pass-through, and
the closed-gate no-transmit path.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
The playback stream was negotiated with a ~256ms (12288-frame) maxsize
buffer. The sink drains the graph quantum (1024 frames) per cycle, so one
of our buffers lasted ~12 cycles and `process` was called only ~4x/sec,
each time asking us to fill all 12288 frames -- far more than the 200ms
(9600-sample) playout ring could ever hold. So ~half of every buffer was
silence-fill: a steady ~46% underrun, audible as constant crackle. This
is a consumer-side buffer-size bug, upstream of production pacing, which
is why earlier mixer-pacing attempts never moved the numbers.
Fix: pass an explicit SPA_TYPE_OBJECT_ParamBuffers param on connect,
pinning buffer size to one 1024-frame quantum (2048 bytes mono S16LE).
PipeWire now hands us a quantum-sized buffer ~47x/sec, the ring satisfies
every callback, and slice.len()/stride equals the quantum so we never
over-pull. A node.latency hint is added too (not load-bearing on its own
-- the hint alone changed nothing; the Buffers param is the fix). Note:
pipewire 0.9.2 only exposes feature v0_3_32, so Buffer::requested() is
unreachable -- pinning the buffer size is the available lever.
Verified with the probe (quantum=1024, 47 cb/s, underrun +0 steady) and
by ear: clean 440Hz tone, no clicks. Local playout path only -- not yet
verified on a live two-peer call.
Also in this commit (the investigation scaffolding that proved it out):
- Fill-paced mixer: production tracks the hardware clock via a shared
exact ring-occupancy gauge (Arc<AtomicUsize>) kept near
PLAYBACK_TARGET_SAMPLES, replacing the fixed 20ms timer that beat
against the 1024 quantum.
- src/bin/audio_probe.rs: drives a sine through the real start_playback
path with no network/mic, for isolating the local output stage.
- playout-health logging: quiet in normal use (logs only on underrun/
dropped > 0); set PEERSPEAK_AUDIO_VERBOSE=1 for the per-second
heartbeat (audio_probe sets it automatically).
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
molluskandClaude Opus 4.8