Files
punktfunk/crates/punktfunk-host/src/audio/linux.rs
T
enricobuehlerandClaude Opus 4.8 3cc3c02b42 feat(gamestream): AV1 negotiation + 5.1/7.1 surround audio
Codec negotiation (M2 polish):
- ServerCodecModeSupport now advertises what we encode: H264|HEVC|AV1_MAIN8
  = 65793 (flags verified against moonlight-common-c Limelight.h). The old
  placeholder 3843 wrongly claimed HEVC Main10 + 4:4:4 and no AV1. Main10
  bits stay off on purpose: Moonlight ties 10-bit to HDR, and capture is
  8-bit SDR BGRx with no HDR metadata path (av1_nvenc -highbitdepth was
  validated working for later).
- RTSP ANNOUNCE: bitStreamFormat 0/1/2 -> H264/HEVC/AV1 (already plumbed to
  av1_nvenc; validated e2e via `m0 --codec av1` + ffprobe av01), and a
  dynamicRangeMode!=0 request now logs + falls back to 8-bit SDR.

Surround audio (M2 polish):
- ANNOUNCE x-nv-audio.surround.{numChannels,AudioQuality} +
  x-nv-aqos.packetDuration -> per-session AudioParams; DESCRIBE advertises
  all six Opus configs (normal before HQ per channel count). Normal-quality
  mappings are pre-rotated for the client's GFE-order LFE swap
  (RtspConnection.c, verified verbatim) so its derived decoder mapping
  equals our encoder mapping — including 7.1, where Sunshine's rotate only
  covers [3,6) and scrambles LFE/SL/SR.
- 5.1/7.1 encode via libopus multistream (audiopus_sys, the sys layer the
  opus crate already links) with Sunshine's layouts/bitrates, RAII wrapper;
  the live-validated stereo wire is byte-identical (plain Opus, no FEC).
- Surround sessions add Sunshine-style RS(4,2) audio FEC (packetType 127 +
  AUDIO_FEC_HEADER, the OpenFEC parity matrix both ends hardcode, nanors
  gemm semantics verified from nanors/rs.c).
- PipeWire capture generalized to the negotiated channel count with explicit
  FL FR FC LFE RL RR [SL SR] positions; missing sink channels are zero-
  filled by the channel-mixer. PwAudioCapturer now tears down cleanly on
  Drop (pipewire channel -> loop quit), so a channel-count change can
  reopen without leaking a capture stream.

Tests: serverinfo mask, RTSP codec/audio param parsing, DESCRIBE contents,
surround-params strings + client-swap round trip, FEC parity self-recovery
and packet layout, real-codec 5.1 channel-identity round trip, and an
ignored live test (ran green against a 6ch null sink monitor).

Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
2026-06-10 15:41:15 +00:00

246 lines
9.4 KiB
Rust

//! PipeWire audio capture of the default sink's monitor (system output).
//!
//! Connects to the user's PipeWire daemon (via `XDG_RUNTIME_DIR`, inherited from the Sway
//! session) and opens an input stream with `stream.capture.sink=true`, which routes the
//! default sink's monitor into us — no portal needed (unlike screen capture). The (`!Send`)
//! MainLoop/Stream live on a dedicated thread; interleaved `f32` chunks leave over a bounded
//! channel (dropped if the encoder falls behind, never blocking the PipeWire loop).
//!
//! The stream is opened at the *session's* channel count (2/6/8). If the sink has fewer
//! channels than requested, PipeWire's channel-mixer fills the extra positions with silence
//! (zero upmix), so a stereo desktop still produces a valid 5.1/7.1 capture. Dropping the
//! capturer quits the loop thread (via a `pipewire::channel` Terminate message), tearing the
//! stream down promptly — required so a surround session can replace a stereo capturer
//! without leaking a PipeWire consumer (see CLAUDE.md: a wedged link head-blocks the daemon).
use super::{AudioCapturer, SAMPLE_RATE};
use anyhow::{anyhow, Context, Result};
use std::sync::mpsc::{sync_channel, Receiver, RecvTimeoutError};
use std::thread;
use std::time::Duration;
/// Message asking the PipeWire loop thread to quit (sent from `Drop`).
struct Terminate;
pub struct PwAudioCapturer {
chunks: Receiver<Vec<f32>>,
channels: u32,
quit: pipewire::channel::Sender<Terminate>,
}
impl PwAudioCapturer {
pub fn open(channels: u32) -> Result<PwAudioCapturer> {
anyhow::ensure!(
matches!(channels, 1 | 2 | 6 | 8),
"unsupported audio channel count {channels} (want 2, 6 or 8)"
);
let (tx, rx) = sync_channel::<Vec<f32>>(64);
let (quit_tx, quit_rx) = pipewire::channel::channel::<Terminate>();
thread::Builder::new()
.name("punktfunk-pw-audio".into())
.spawn(move || {
if let Err(e) = pw_thread(tx, quit_rx, channels) {
tracing::error!(error = %format!("{e:#}"), "pipewire audio thread failed");
}
})
.context("spawn pipewire audio thread")?;
Ok(PwAudioCapturer {
chunks: rx,
channels,
quit: quit_tx,
})
}
}
impl Drop for PwAudioCapturer {
fn drop(&mut self) {
// Ask the loop thread to quit; the stream/core/loop unwind there (RAII). A failed
// send means the thread already exited — nothing to tear down.
let _ = self.quit.send(Terminate);
}
}
impl AudioCapturer for PwAudioCapturer {
fn next_chunk(&mut self) -> Result<Vec<f32>> {
match self.chunks.recv_timeout(Duration::from_secs(5)) {
Ok(c) => Ok(c),
Err(RecvTimeoutError::Timeout) => Err(anyhow!("no PipeWire audio within 5s")),
Err(RecvTimeoutError::Disconnected) => Err(anyhow!("pipewire audio thread ended")),
}
}
fn channels(&self) -> u32 {
self.channels
}
fn drain(&mut self) {
while self.chunks.try_recv().is_ok() {}
}
}
/// SPA channel position array for the GameStream surround order FL FR FC LFE RL RR [SL SR]
/// (= the PipeWire/PulseAudio default map for 6/8 channels, and the order Moonlight's
/// renderers expect — moonlight-common-c: "we use FL FR C LFE RL RR SL SR"). Values are
/// `enum spa_audio_channel` (spa/param/audio/raw.h): FL=3 FR=4 FC=5 LFE=6 SL=7 SR=8 RL=12
/// RR=13.
fn spa_positions(channels: u32) -> [u32; 64] {
const FL: u32 = 3;
const FR: u32 = 4;
const FC: u32 = 5;
const LFE: u32 = 6;
const SL: u32 = 7;
const SR: u32 = 8;
const RL: u32 = 12;
const RR: u32 = 13;
const MONO: u32 = 2;
let mut pos = [0u32; 64];
let order: &[u32] = match channels {
1 => &[MONO],
2 => &[FL, FR],
6 => &[FL, FR, FC, LFE, RL, RR],
8 => &[FL, FR, FC, LFE, RL, RR, SL, SR],
_ => unreachable!("validated in open()"),
};
pos[..order.len()].copy_from_slice(order);
pos
}
fn pw_thread(
tx: std::sync::mpsc::SyncSender<Vec<f32>>,
quit_rx: pipewire::channel::Receiver<Terminate>,
channels: u32,
) -> Result<()> {
use pipewire as pw;
use pw::{properties::properties, spa};
use spa::param::audio::{AudioFormat, AudioInfoRaw};
use spa::pod::Pod;
crate::pwinit::ensure_init();
let mainloop = pw::main_loop::MainLoopRc::new(None).context("pw audio MainLoop")?;
let context = pw::context::ContextRc::new(&mainloop, None).context("pw audio Context")?;
let core = context
.connect_rc(None)
.context("pw audio connect (is PipeWire running in this session?)")?;
// Cross-thread teardown: the capturer's Drop sends Terminate; quit the loop here.
let _quit_guard = quit_rx.attach(mainloop.loop_(), {
let mainloop = mainloop.clone();
move |_| mainloop.quit()
});
let stream = pw::stream::StreamBox::new(
&core,
"punktfunk-audio",
properties! {
*pw::keys::MEDIA_TYPE => "Audio",
*pw::keys::MEDIA_CATEGORY => "Capture",
*pw::keys::MEDIA_ROLE => "Music",
// Capture the default sink's monitor (system output), not a microphone.
*pw::keys::STREAM_CAPTURE_SINK => "true",
// Ask for a ~5ms quantum (= one Opus frame) so buffers arrive smoothly rather than
// in large bursts the client's low-latency jitter buffer would hear as glitching.
*pw::keys::NODE_LATENCY => "240/48000",
},
)
.context("pw audio Stream")?;
let _listener = stream
.add_local_listener_with_user_data(tx)
.state_changed(|_s, _ud, old, new| {
tracing::info!(?old, ?new, "pipewire audio stream state");
})
.param_changed(|_stream, _tx, id, param| {
let Some(param) = param else { return };
if id != pw::spa::param::ParamType::Format.as_raw() {
return;
}
let mut info = AudioInfoRaw::default();
if info.parse(param).is_ok() {
tracing::info!(
format = ?info.format(),
rate = info.rate(),
channels = info.channels(),
"audio format negotiated"
);
}
})
.process(|stream, tx| {
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 d = &mut datas[0];
let (offset, size) = {
let c = d.chunk();
(c.offset() as usize, c.size() as usize)
};
let Some(buf) = d.data() else { return };
if offset > buf.len() {
return;
}
let region = &buf[offset..(offset + size).min(buf.len())];
// Negotiated as F32LE; reinterpret the byte region as interleaved f32.
let n = region.len() / 4;
static FIRST: std::sync::atomic::AtomicBool =
std::sync::atomic::AtomicBool::new(true);
if FIRST.swap(false, std::sync::atomic::Ordering::Relaxed) {
tracing::info!(samples = n, "audio first capture buffer");
}
let mut samples = Vec::with_capacity(n);
for i in 0..n {
let b = [
region[i * 4],
region[i * 4 + 1],
region[i * 4 + 2],
region[i * 4 + 3],
];
samples.push(f32::from_le_bytes(b));
}
let _ = tx.try_send(samples); // drop if the encoder is behind
}));
if outcome.is_err() {
tracing::error!("panic in pipewire audio callback — chunk dropped");
}
})
.register()
.context("register audio stream listener")?;
// Request F32LE, 48 kHz, at the session's channel count with explicit positions —
// PipeWire's channel-mixer up/downmixes the sink monitor to this layout.
let mut info = AudioInfoRaw::new();
info.set_format(AudioFormat::F32LE);
info.set_rate(SAMPLE_RATE);
info.set_channels(channels);
info.set_position(spa_positions(channels));
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 audio format pod")?
.0
.into_inner();
let mut params = [Pod::from_bytes(&values).context("audio pod from bytes")?];
stream
.connect(
spa::utils::Direction::Input,
None, // PW_ID_ANY — autoconnect to the default sink monitor
pw::stream::StreamFlags::AUTOCONNECT | pw::stream::StreamFlags::MAP_BUFFERS,
&mut params,
)
.context("pw audio stream connect")?;
mainloop.run();
tracing::debug!("pipewire audio loop exited (capturer dropped)");
Ok(())
}