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punktfunk/crates/pf-client-core/src/audio.rs
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fix(client/audio): the PipeWire callback stops filling the buffer ceiling every cycle
The playback process callback sized its writes from the mapped buffer's
capacity — PipeWire's quantum-limit, 8192 frames ≈ 170 ms — instead of
the graph's per-cycle ask (pw_buffer.requested). Every cycle therefore
queued up to 170 ms of PCM downstream of the ring, and, worse, taught
JitterPolicy that the device drains 170 ms per callback: the underrun
floor (want + one frame) rose above any depth the A/V sync loop may
request, so sync measured audio ~280 ms late and was forbidden — by its
own continuity rule — from draining it. The first on-glass run of the
latency overhaul showed exactly that: audio buffer 272 ms, a/v +284 ms,
stable.

Honor requested (capacity remains both the ceiling and the fallback for
requested == 0), and log requested-vs-capacity once per stream in the
shape of the host's per-capture-open quantum line, so the next on-glass
report can say which one is sizing the writes.

Needs libpipewire >= 0.3.49 (2022-03) for the requested field; every
ship target clears that.

Verified on .21: cargo clippy -p pf-client-core --all-targets -D
warnings clean, 167 tests pass, fmt clean.
2026-08-08 02:04:15 +02:00

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//! Audio: playback (decoded PCM → a PipeWire playback stream) and the microphone uplink
//! (PipeWire capture → Opus → 0xCB datagrams, the inverse of the host's virtual mic).
//!
//! Playback mirrors the host's virtual-mic producer (`punktfunk-host::audio::linux`) with
//! the same adaptive jitter buffer: the session pump pushes 5 ms Opus-decoded chunks on
//! the network clock; PipeWire pulls whole quanta on the device clock. Prime to ~3
//! quanta before producing, cap the ring so latency stays bounded, re-prime after a real
//! drain.
use anyhow::{Context, Result};
use punktfunk_core::client::NativeClient;
use std::collections::VecDeque;
use std::sync::mpsc::{Receiver, SyncSender, TrySendError};
use std::sync::Arc;
const SAMPLE_RATE: u32 = 48_000;
/// Mic capture is MONO: voice is mono at the source, the host accepts any Opus channel
/// layout (its stereo decoder upmixes), and half the samples halve the encode + wire cost.
const MIC_CHANNELS: usize = 1;
/// Mic frames are 10 ms (480 mono samples) — any size ≤ 120 ms is fine host-side; 10 ms
/// halves the frame-fill share of mouth-to-ear latency vs the old 20 ms.
const MIC_FRAME: usize = 480;
struct Terminate;
/// A selectable PipeWire endpoint for the settings pickers.
#[derive(Clone, Debug)]
pub struct AudioDevice {
/// `node.name` — the stable key the streams target via `target.object`.
pub name: String,
/// `node.description` — the human label the picker shows.
pub description: String,
}
/// Enumerate audio endpoints: `(sinks, sources)`. One registry roundtrip on a private
/// mainloop (a few ms against a live PipeWire); no daemon errors out and the caller
/// simply shows no pickers.
pub fn devices() -> Result<(Vec<AudioDevice>, Vec<AudioDevice>)> {
use pipewire as pw;
use std::cell::RefCell;
use std::rc::Rc;
static PW_INIT: std::sync::Once = std::sync::Once::new();
PW_INIT.call_once(pw::init);
let mainloop = pw::main_loop::MainLoopRc::new(None).context("pw MainLoop")?;
let context = pw::context::ContextRc::new(&mainloop, None).context("pw Context")?;
let core = context
.connect_rc(None)
.context("pw connect (is PipeWire running in this session?)")?;
let registry = core.get_registry_rc().context("pw registry")?;
let found: Rc<RefCell<(Vec<AudioDevice>, Vec<AudioDevice>)>> = Rc::default();
let _reg_listener = registry
.add_listener_local()
.global({
let found = found.clone();
move |g| {
let Some(props) = g.props else { return };
let sink = match props.get("media.class") {
Some("Audio/Sink") => true,
Some("Audio/Source") => false,
_ => return,
};
let Some(name) = props.get("node.name") else {
return;
};
let description = props
.get("node.description")
.or_else(|| props.get("node.nick"))
.unwrap_or(name)
.to_string();
let dev = AudioDevice {
name: name.to_string(),
description,
};
let mut f = found.borrow_mut();
if sink { &mut f.0 } else { &mut f.1 }.push(dev);
}
})
.register();
// The registry replays existing globals asynchronously; one core sync marks the
// point they've all been delivered — quit the loop there.
let pending = core.sync(0).context("pw sync")?;
let _core_listener = core
.add_listener_local()
.done({
let mainloop = mainloop.clone();
move |_, seq| {
if seq == pending {
mainloop.quit();
}
}
})
.register();
mainloop.run();
let result = found.borrow().clone();
Ok(result)
}
pub struct AudioPlayer {
pcm_tx: SyncSender<Vec<f32>>,
/// Drained chunk Vecs coming back from the PipeWire consumer for reuse (the pool half
/// of the pcm channel — see [`AudioPlayer::take_buffer`]).
recycle_rx: Receiver<Vec<f32>>,
quit_tx: pipewire::channel::Sender<Terminate>,
thread: Option<std::thread::JoinHandle<()>>,
/// A/V sync hand-off with the PipeWire callback: it publishes the ring depth, the decode
/// thread posts the depth the sync loop wants. See [`punktfunk_core::audio::AudioSyncCell`].
sync: Arc<punktfunk_core::audio::AudioSyncCell>,
}
impl AudioPlayer {
/// Spawn the PipeWire playback thread for `channels` (2/6/8, canonical wire order
/// FL FR FC LFE RL RR SL SR). Failure (no PipeWire in the session) is survivable — the
/// caller streams video-only.
pub fn spawn(channels: u32) -> Result<AudioPlayer> {
// 64 × 5 ms = 320 ms of slack between the pump and the PipeWire loop.
let (pcm_tx, pcm_rx) = std::sync::mpsc::sync_channel::<Vec<f32>>(64);
// Return path: the process callback sends each drained Vec back for reuse, so
// steady-state playback stops allocating (~200 chunks/s otherwise). Same capacity
// as the data channel; a full pool just drops the Vec (plain deallocation).
let (recycle_tx, recycle_rx) = std::sync::mpsc::sync_channel::<Vec<f32>>(64);
let (quit_tx, quit_rx) = pipewire::channel::channel::<Terminate>();
let sync: Arc<punktfunk_core::audio::AudioSyncCell> = Arc::default();
let sync_cb = sync.clone();
let thread = std::thread::Builder::new()
.name("punktfunk-audio".into())
.spawn(move || {
if let Err(e) = pw_thread(pcm_rx, recycle_tx, quit_rx, channels as usize, sync_cb) {
tracing::warn!(error = %e, "audio playback thread ended");
}
})
.context("spawn audio thread")?;
Ok(AudioPlayer {
pcm_tx,
recycle_rx,
quit_tx,
thread: Some(thread),
sync,
})
}
/// The A/V sync hand-off cell — the decode thread reads the ring depth from it and posts the
/// depth the sync loop wants back through it.
pub fn sync_cell(&self) -> Arc<punktfunk_core::audio::AudioSyncCell> {
self.sync.clone()
}
/// A recycled chunk Vec from the pool, empty but with its capacity intact — fill it
/// and hand it back through [`push`](Self::push). Allocates only when the pool is dry
/// (startup, or after the PipeWire side dropped chunks).
pub fn take_buffer(&self) -> Vec<f32> {
self.recycle_rx.try_recv().unwrap_or_default()
}
/// Queue one interleaved f32 chunk (in the session's channel layout). Drops the chunk if the
/// PipeWire side is wedged (the renderer conceals the gap; never block the session pump).
pub fn push(&self, pcm: Vec<f32>) {
if let Err(TrySendError::Disconnected(_)) = self.pcm_tx.try_send(pcm) {
// Thread already dead — Drop will reap it; nothing to do per-chunk.
}
}
}
impl Drop for AudioPlayer {
fn drop(&mut self) {
let _ = self.quit_tx.send(Terminate);
if let Some(t) = self.thread.take() {
let _ = t.join();
}
}
}
/// Producer-side state: incoming decoded PCM and the ring the process callback drains.
struct PlayerData {
rx: Receiver<Vec<f32>>,
/// Drained chunk Vecs go back here for the decode side to refill (allocation pool).
recycle: SyncSender<Vec<f32>>,
ring: VecDeque<f32>,
/// Shared ms-denominated de-jitter policy: prime depth, drift correction, de-prime
/// hysteresis. Replaces the old `3 × quantum` target, which meant 15 ms at a 5 ms graph
/// quantum and a silent 64 ms at a 20 ms one, and the `if ring.is_empty()` re-prime, where
/// one transient drain manufactured a whole target's worth of fresh silence.
policy: punktfunk_core::audio::JitterPolicy,
/// Interleaved channel count this stream was opened with (2/6/8).
channels: usize,
/// Diagnostics (WP0.3), logged ~every 10 s: the audio plane used to be entirely silent in a
/// client log, so a latency or dropout report had nothing to go on.
underruns: u64,
sheds: u64,
callbacks: u64,
/// A/V sync hand-off with the decode thread (depth out, target in).
sync: Arc<punktfunk_core::audio::AudioSyncCell>,
}
fn pw_thread(
pcm_rx: Receiver<Vec<f32>>,
recycle_tx: SyncSender<Vec<f32>>,
quit_rx: pipewire::channel::Receiver<Terminate>,
channels: usize,
sync: Arc<punktfunk_core::audio::AudioSyncCell>,
) -> Result<()> {
use pipewire as pw;
use pw::{properties::properties, spa};
use spa::param::audio::{AudioFormat, AudioInfoRaw};
use spa::pod::Pod;
static PW_INIT: std::sync::Once = std::sync::Once::new();
PW_INIT.call_once(pw::init);
let mainloop = pw::main_loop::MainLoopRc::new(None).context("pw MainLoop")?;
let context = pw::context::ContextRc::new(&mainloop, None).context("pw Context")?;
let core = context
.connect_rc(None)
.context("pw connect (is PipeWire running in this session?)")?;
let _quit_guard = quit_rx.attach(mainloop.loop_(), {
let mainloop = mainloop.clone();
move |_| mainloop.quit()
});
let mut props = properties! {
*pw::keys::MEDIA_TYPE => "Audio",
*pw::keys::MEDIA_CATEGORY => "Playback",
*pw::keys::MEDIA_ROLE => "Game",
*pw::keys::NODE_NAME => "punktfunk-client",
*pw::keys::NODE_DESCRIPTION => "Punktfunk Stream",
// ~5 ms quantum (one Opus frame) keeps the ring — and so the latency — small.
*pw::keys::NODE_LATENCY => "240/48000",
};
// The Settings speaker pick (session main maps `Settings::speaker_device` here);
// unset/empty = PipeWire's default routing.
if let Ok(target) = std::env::var("PUNKTFUNK_AUDIO_SINK") {
if !target.is_empty() {
// Raw key: the `keys::TARGET_OBJECT` constant is feature-gated on a newer
// libpipewire than we require; the wire name is stable.
props.insert("target.object", target);
}
}
let stream =
pw::stream::StreamBox::new(&core, "punktfunk-client", props).context("pw Stream")?;
let ud = PlayerData {
rx: pcm_rx,
recycle: recycle_tx,
ring: VecDeque::new(),
policy: punktfunk_core::audio::JitterPolicy::new(
punktfunk_core::audio::JitterTuning::PIPEWIRE,
channels as u8,
),
channels,
underruns: 0,
sheds: 0,
callbacks: 0,
sync,
};
let _listener = stream
.add_local_listener_with_user_data(ud)
.state_changed(|_s, _ud, old, new| {
tracing::debug!(?old, ?new, "pipewire playback stream state");
})
.process(|stream, ud| {
let outcome = std::panic::catch_unwind(std::panic::AssertUnwindSafe(|| {
let Some(mut buffer) = stream.dequeue_buffer() else {
return;
};
while let Ok(mut chunk) = ud.rx.try_recv() {
ud.ring.extend(chunk.iter().copied());
// Return the drained Vec to the pool; a full/closed pool drops it.
chunk.clear();
let _ = ud.recycle.try_send(chunk);
}
// The graph asks for `requested` frames this cycle (one quantum, after
// rate-matching); the mapped buffer is sized for the WORST case — PipeWire's
// `quantum-limit`, 8192 frames ≈ 170 ms — not for this cycle. Filling to
// capacity queued ~170 ms per buffer downstream of the ring and, worse, taught
// the jitter policy that the device drains 170 ms per callback, which lifted
// the underrun floor (`want` + one frame) above any depth the A/V sync loop is
// allowed to ask for: audio sat a stable ~270 ms late and, by the continuity
// rule, sync was FORBIDDEN from draining it. Capacity is only the ceiling;
// `requested == 0` (no adapter suggestion) falls back to it.
let requested = usize::try_from(buffer.requested()).unwrap_or(0);
let stride = 4 * ud.channels; // F32LE interleaved
let datas = buffer.datas_mut();
if datas.is_empty() {
return;
}
let data = &mut datas[0];
let max_frames = data.data().map(|s| s.len() / stride).unwrap_or(0);
let want_frames = if requested > 0 {
requested.min(max_frames)
} else {
max_frames
};
let want = want_frames * ud.channels;
// Once per stream, in the shape of the host's per-capture-open quantum log:
// whether the graph's request or the buffer ceiling is sizing our writes is
// exactly what an on-glass latency report needs to say.
if ud.callbacks == 0 {
tracing::info!(
requested_frames = requested,
capacity_frames = max_frames,
write_frames = want_frames,
write_ms = want_frames / 48,
"audio playback quantum"
);
}
// A/V sync: take whatever depth the decode thread's sync loop last asked for, and
// publish where the ring actually is so it can measure the result. The policy
// clamps the request between its own underrun floor and the hard cap — continuity
// outranks sync, always (see `JitterPolicy::set_sync_target`).
ud.policy.set_sync_target(ud.sync.target());
ud.sync.publish_depth(ud.ring.len());
// Shared de-jitter policy: prime depth in MILLISECONDS, smooth drift correction
// (a crossfaded 5 ms shed) so latency returns to target instead of ratcheting,
// and a hard cap as the backstop.
let step = ud.policy.step(ud.ring.len(), want);
if step.drop_front > 0 {
ud.sheds += 1;
punktfunk_core::audio::crossfade_drop(
&mut ud.ring,
step.drop_front,
step.crossfade,
);
}
let mut ran_short = false;
let n_frames = if let Some(slice) = data.data() {
for k in 0..want {
let s = if step.silence {
0.0
} else {
ud.ring.pop_front().unwrap_or_else(|| {
ran_short = true;
0.0
})
};
let off = k * 4;
slice[off..off + 4].copy_from_slice(&s.to_le_bytes());
}
want_frames
} else {
0
};
// No-op while un-primed (the policy ignores it), so a deliberate priming silence
// is never miscounted as an underrun.
ud.policy.note_read(ran_short);
ud.underruns += u64::from(ran_short);
ud.callbacks += 1;
// ~10 s at a 5 ms quantum; the exact cadence does not matter, only that the
// plane stops being invisible.
if ud.callbacks % 2_000 == 0 {
tracing::debug!(
buffer_ms = ud.policy.avg_depth_ms(),
target_ms = ud.policy.target_ms(),
underruns = ud.underruns,
drift_sheds = ud.sheds,
"audio playback"
);
}
let chunk = data.chunk_mut();
*chunk.offset_mut() = 0;
*chunk.stride_mut() = stride as _;
*chunk.size_mut() = (stride * n_frames) as _;
}));
if outcome.is_err() {
tracing::error!("panic in pipewire playback callback");
}
})
.register()
.context("register playback listener")?;
let mut info = AudioInfoRaw::new();
info.set_format(AudioFormat::F32LE);
info.set_rate(SAMPLE_RATE);
info.set_channels(channels as u32);
// Channel positions in canonical wire order (FL FR FC LFE RL RR SL SR) so PipeWire routes each
// slot to the matching speaker (and downmixes when the sink has fewer). Identity, no permute.
let order = punktfunk_core::audio::spa_positions(channels as u8);
let mut positions = [0u32; 64];
positions[..order.len()].copy_from_slice(order);
info.set_position(positions);
let obj = pw::spa::pod::Object {
type_: pw::spa::utils::SpaTypes::ObjectParamFormat.as_raw(),
id: pw::spa::param::ParamType::EnumFormat.as_raw(),
properties: info.into(),
};
let values: Vec<u8> = pw::spa::pod::serialize::PodSerializer::serialize(
std::io::Cursor::new(Vec::new()),
&pw::spa::pod::Value::Object(obj),
)
.context("serialize format pod")?
.0
.into_inner();
let mut params = [Pod::from_bytes(&values).context("pod from bytes")?];
stream
.connect(
spa::utils::Direction::Output,
None,
pw::stream::StreamFlags::AUTOCONNECT | pw::stream::StreamFlags::MAP_BUFFERS,
&mut params,
)
.context("pw stream connect")?;
mainloop.run();
tracing::debug!("pipewire playback loop exited");
Ok(())
}
/// The microphone uplink: capture the default input device (or the picked / echo-cancelled
/// source), Opus-encode 10 ms mono chunks, ship them as 0xCB datagrams into the host's
/// virtual PipeWire source.
pub struct MicStreamer {
quit_tx: pipewire::channel::Sender<Terminate>,
thread: Option<std::thread::JoinHandle<()>>,
}
impl MicStreamer {
/// `muted` is the in-stream mute (B4), shared live with the capture callback: set, the
/// callback keeps pulling and discarding whole frames but sends nothing. Muting by
/// STOPPING the stream was rejected — it re-primes the device buffers and re-runs the
/// source selection below on every unmute, so the first second back is glitchy.
///
/// `echo_cancel` is the Settings toggle; `PUNKTFUNK_NO_AEC=1` overrides it off.
pub fn spawn(
connector: Arc<NativeClient>,
muted: Arc<std::sync::atomic::AtomicBool>,
echo_cancel: bool,
) -> Result<MicStreamer> {
let (quit_tx, quit_rx) = pipewire::channel::channel::<Terminate>();
let thread = std::thread::Builder::new()
.name("punktfunk-mic".into())
.spawn(move || {
if let Err(e) = mic_thread(&connector, quit_rx, muted, echo_cancel) {
tracing::warn!(error = %e, "mic uplink thread ended");
}
})
.context("spawn mic thread")?;
Ok(MicStreamer {
quit_tx,
thread: Some(thread),
})
}
}
impl Drop for MicStreamer {
fn drop(&mut self) {
let _ = self.quit_tx.send(Terminate);
if let Some(t) = self.thread.take() {
let _ = t.join();
}
}
}
/// Capture-side state: accumulated PCM and the Opus encoder (encoding a 10 ms frame is
/// well under 100 µs — fine inside the process callback).
struct MicData {
connector: Arc<NativeClient>,
ring: VecDeque<f32>,
encoder: opus::Encoder,
seq: u32,
out: Vec<u8>,
/// The in-stream mute (B4), flipped by the session's chord. Read per callback.
muted: Arc<std::sync::atomic::AtomicBool>,
}
/// Whether the mic echo-cancellation hooks run this session: the `echo_cancel` setting, with
/// `PUNKTFUNK_NO_AEC=1` as a one-way override OFF. The env var wins — it is the escape hatch
/// for a box whose canceller misbehaves, and it predates the setting; nothing turns AEC back
/// on once it is set. Here the hook is the echo-cancelled-source preference below; the WASAPI
/// twin gates its Communications stream category the same way.
fn aec_enabled(echo_cancel: bool) -> bool {
echo_cancel && !std::env::var("PUNKTFUNK_NO_AEC").is_ok_and(|v| !v.is_empty() && v != "0")
}
/// The capture stream's `target.object`, in preference order: the Settings microphone pick
/// (`Settings::mic_device` via session main's `PUNKTFUNK_AUDIO_SOURCE`) verbatim, else — so a
/// desktop that already runs `module-echo-cancel` stops feeding its own downlink audio back
/// into the host's virtual mic — the first echo-cancelled source in the graph. `None` = the
/// user picked nothing and no such source exists: PipeWire's default routing, as before.
///
/// Preference-only by design: loading `libpipewire-module-echo-cancel` ourselves needs
/// `pw_context_load_module`, which the pipewire crate (0.9) doesn't expose safely — until it
/// does, we only ever target processing the user (or their session) already set up.
fn mic_capture_target(echo_cancel: bool) -> Option<String> {
if let Ok(target) = std::env::var("PUNKTFUNK_AUDIO_SOURCE") {
if !target.is_empty() {
return Some(target);
}
}
if !aec_enabled(echo_cancel) {
return None;
}
let name = echo_cancel_source()?;
tracing::info!(
source = %name,
"mic capture targets the echo-cancelled source (Echo cancellation off, or \
PUNKTFUNK_NO_AEC=1, disables this)"
);
Some(name)
}
/// Find an existing echo-cancelled capture node: the first `Audio/Source` whose `node.name`
/// or description says echo-cancel (`module-echo-cancel`'s convention — `echo-cancel-*`
/// nodes, "Echo-Cancel …" descriptions; PulseAudio-compat setups match too). One registry
/// roundtrip via [`devices`]; any failure reads as "none".
fn echo_cancel_source() -> Option<String> {
let (_, sources) = devices().ok()?;
sources.into_iter().find_map(|d| {
let name = d.name.to_ascii_lowercase();
let desc = d.description.to_ascii_lowercase();
(name.contains("echo-cancel")
|| name.contains("echo_cancel")
|| desc.contains("echo-cancel")
|| desc.contains("echo cancel"))
.then_some(d.name)
})
}
fn mic_thread(
connector: &Arc<NativeClient>,
quit_rx: pipewire::channel::Receiver<Terminate>,
muted: Arc<std::sync::atomic::AtomicBool>,
echo_cancel: bool,
) -> Result<()> {
use pipewire as pw;
use pw::{properties::properties, spa};
use spa::param::audio::{AudioFormat, AudioInfoRaw};
use spa::pod::Pod;
static PW_INIT: std::sync::Once = std::sync::Once::new();
PW_INIT.call_once(pw::init);
let mut encoder =
opus::Encoder::new(SAMPLE_RATE, opus::Channels::Mono, opus::Application::Voip)
.map_err(|e| anyhow::anyhow!("opus encoder: {e}"))?;
// Voice tuning: 48 kbps mono is transparent for speech; in-band FEC + an assumed 10 %
// loss let the host's decoder rebuild a lost 0xCB datagram from its successor instead
// of concealing (datagrams are fire-and-forget — this FEC is the only redundancy).
let _ = encoder.set_bitrate(opus::Bitrate::Bits(48_000));
let _ = encoder.set_inband_fec(true);
let _ = encoder.set_packet_loss_perc(10);
let mainloop = pw::main_loop::MainLoopRc::new(None).context("pw mic MainLoop")?;
let context = pw::context::ContextRc::new(&mainloop, None).context("pw mic Context")?;
let core = context
.connect_rc(None)
.context("pw mic connect (is PipeWire running in this session?)")?;
let _quit_guard = quit_rx.attach(mainloop.loop_(), {
let mainloop = mainloop.clone();
move |_| mainloop.quit()
});
let mut props = properties! {
*pw::keys::MEDIA_TYPE => "Audio",
*pw::keys::MEDIA_CATEGORY => "Capture",
*pw::keys::MEDIA_ROLE => "Communication",
*pw::keys::NODE_NAME => "punktfunk-mic-capture",
*pw::keys::NODE_DESCRIPTION => "Punktfunk Microphone",
// ~10 ms quantum (one mic frame). Without it the capture stream inherits the graph
// quantum — commonly 10242048 samples, so the mic arrived in 2143 ms bursts that
// sat ahead of the encoder as latency (the playback stream always asked for 5 ms).
*pw::keys::NODE_LATENCY => "480/48000",
};
if let Some(target) = mic_capture_target(echo_cancel) {
// Raw key: the `keys::TARGET_OBJECT` constant is feature-gated on a newer
// libpipewire than we require; the wire name is stable.
props.insert("target.object", target);
}
let stream = pw::stream::StreamBox::new(&core, "punktfunk-mic-capture", props)
.context("pw mic Stream")?;
let ud = MicData {
connector: connector.clone(),
ring: VecDeque::new(),
encoder,
seq: 0,
out: vec![0u8; 4000],
muted,
};
let _listener = stream
.add_local_listener_with_user_data(ud)
.state_changed(|_s, _ud, old, new| {
tracing::debug!(?old, ?new, "pipewire mic capture stream state");
})
.process(|stream, ud| {
let outcome = std::panic::catch_unwind(std::panic::AssertUnwindSafe(|| {
let Some(mut buffer) = stream.dequeue_buffer() else {
return;
};
let datas = buffer.datas_mut();
if datas.is_empty() {
return;
}
let data = &mut datas[0];
let n = data.chunk().size() as usize;
if let Some(slice) = data.data() {
for s in slice[..n.min(slice.len())].chunks_exact(4) {
ud.ring
.push_back(f32::from_le_bytes([s[0], s[1], s[2], s[3]]));
}
}
// Muted (B4): the stream stays open and the device keeps its primed buffers —
// only the sending stops. Whole frames are discarded so the ring can't grow,
// and `seq` deliberately does NOT advance: the host sees one continuous
// sequence with a silent pause in the middle rather than a gap the size of the
// mute, which its de-jitter would try to conceal frame by frame.
if ud.muted.load(std::sync::atomic::Ordering::Relaxed) {
let whole =
(ud.ring.len() / (MIC_FRAME * MIC_CHANNELS)) * (MIC_FRAME * MIC_CHANNELS);
ud.ring.drain(..whole);
return;
}
// Ship every complete 10 ms mono frame.
while ud.ring.len() >= MIC_FRAME * MIC_CHANNELS {
let pcm: Vec<f32> = ud.ring.drain(..MIC_FRAME * MIC_CHANNELS).collect();
match ud.encoder.encode_float(&pcm, &mut ud.out) {
Ok(len) => {
let pts = std::time::SystemTime::now()
.duration_since(std::time::UNIX_EPOCH)
.map(|d| d.as_nanos() as u64)
.unwrap_or(0);
let _ = ud.connector.send_mic(ud.seq, pts, ud.out[..len].to_vec());
ud.seq = ud.seq.wrapping_add(1);
}
Err(e) => tracing::debug!(error = %e, "opus mic encode"),
}
}
}));
if outcome.is_err() {
tracing::error!("panic in pipewire mic callback");
}
})
.register()
.context("register mic listener")?;
let mut info = AudioInfoRaw::new();
info.set_format(AudioFormat::F32LE);
info.set_rate(SAMPLE_RATE);
// Mono: the stream's adapter downmixes whatever layout the source really has.
info.set_channels(MIC_CHANNELS as u32);
let obj = pw::spa::pod::Object {
type_: pw::spa::utils::SpaTypes::ObjectParamFormat.as_raw(),
id: pw::spa::param::ParamType::EnumFormat.as_raw(),
properties: info.into(),
};
let values: Vec<u8> = pw::spa::pod::serialize::PodSerializer::serialize(
std::io::Cursor::new(Vec::new()),
&pw::spa::pod::Value::Object(obj),
)
.context("serialize mic format pod")?
.0
.into_inner();
let mut params = [Pod::from_bytes(&values).context("mic pod from bytes")?];
stream
.connect(
spa::utils::Direction::Input,
None,
pw::stream::StreamFlags::AUTOCONNECT | pw::stream::StreamFlags::MAP_BUFFERS,
&mut params,
)
.context("pw mic stream connect")?;
mainloop.run();
tracing::debug!("pipewire mic capture loop exited");
Ok(())
}