feat(host/windows): WASAPI loopback audio capture
apple / swift (push) Successful in 53s
android / android (push) Failing after 1m59s
ci / rust (push) Failing after 58s
ci / web (push) Successful in 29s
ci / docs-site (push) Successful in 29s
ci / bench (push) Failing after 1m7s
decky / build-publish (push) Successful in 11s
docker / build-push (--build-arg FEDORA_VERSION=44, ci, ci/fedora-rpm.Dockerfile, punktfunk-fedora44-rpm) (push) Successful in 5s
docker / build-push (., web/Dockerfile, punktfunk-web) (push) Successful in 5s
docker / build-push (ci, ci/fedora-rpm.Dockerfile, punktfunk-fedora-rpm) (push) Successful in 4s
docker / build-push (ci, ci/rust-ci.Dockerfile, punktfunk-rust-ci) (push) Successful in 4s
docker / build-push (docs-site, docs-site/Dockerfile, punktfunk-docs) (push) Successful in 3s
flatpak / build-publish (push) Failing after 1s
deb / build-publish (push) Successful in 2m43s
rpm / build-publish (bazzite, punktfunk-fedora-rpm) (push) Failing after 1m32s
rpm / build-publish (fedora-44, punktfunk-fedora44-rpm) (push) Failing after 2m49s
docker / deploy-docs (push) Successful in 18s

Windows AudioCapturer via the wasapi crate (0.23): loopback the default render endpoint (Render device + Direction::Capture + shared mode => STREAMFLAGS_LOOPBACK) at 48 kHz stereo f32 with autoconvert, feeding the existing Opus path with no resampling. Dedicated COM-MTA thread owns the !Send WASAPI objects; interleaved f32 chunks leave over a bounded lossy channel; RAII Drop stops + joins. Bring-up handshake reports a missing endpoint as Err so a session continues without audio. open_audio_capture Windows factory arm + module. Init chain validated live on the VM (open succeeds; next_chunk waits on a silent system). Virtual mic deferred (no Windows virtual-audio endpoint). m3 audio_thread wiring + opus hoist land with the integration task.

Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
This commit is contained in:
2026-06-15 00:57:19 +00:00
parent cbbeaa5c29
commit 45e5157091
4 changed files with 214 additions and 2 deletions
+2
View File
@@ -120,3 +120,5 @@ windows = { version = "0.62", features = [
# Software H.264 encoder (GPU-less path + NVENC fallback). The default `source` feature statically
# compiles OpenH264 (BSD-2) — no system lib, builds on MSVC; nasm on PATH adds the SIMD fast path.
openh264 = "0.9"
# WASAPI loopback audio capture (default render endpoint -> 48 kHz stereo f32 for the Opus path).
wasapi = "0.23"
+9 -2
View File
@@ -37,9 +37,14 @@ pub fn open_audio_capture(channels: u32) -> Result<Box<dyn AudioCapturer>> {
linux::PwAudioCapturer::open(channels).map(|c| Box::new(c) as Box<dyn AudioCapturer>)
}
#[cfg(not(target_os = "linux"))]
#[cfg(target_os = "windows")]
pub fn open_audio_capture(channels: u32) -> Result<Box<dyn AudioCapturer>> {
wasapi_cap::WasapiLoopbackCapturer::open(channels).map(|c| Box::new(c) as Box<dyn AudioCapturer>)
}
#[cfg(not(any(target_os = "linux", target_os = "windows")))]
pub fn open_audio_capture(_channels: u32) -> Result<Box<dyn AudioCapturer>> {
anyhow::bail!("audio capture requires Linux + PipeWire")
anyhow::bail!("audio capture requires Linux + PipeWire or Windows + WASAPI")
}
/// The inverse of [`AudioCapturer`]: a virtual microphone the host *produces*. It registers a
@@ -71,3 +76,5 @@ pub fn open_virtual_mic(_channels: u32) -> Result<Box<dyn VirtualMic>> {
#[cfg(target_os = "linux")]
mod linux;
#[cfg(target_os = "windows")]
mod wasapi_cap;
@@ -0,0 +1,189 @@
//! WASAPI loopback capture of the default render endpoint (system output) — the Windows analogue
//! of the PipeWire sink-monitor backend. Delivers interleaved f32 PCM at 48 kHz stereo, ready for
//! the existing Opus path with NO resampling (WASAPI shared-mode autoconvert does any SRC). WASAPI
//! objects are COM-apartment-bound and not `Send`, so they live on a dedicated thread (mirrors
//! `linux::PwAudioCapturer`); only the channel + stop flag + join handle are in the struct.
use super::{AudioCapturer, SAMPLE_RATE};
use anyhow::{anyhow, Context, Result};
use std::collections::VecDeque;
use std::sync::atomic::{AtomicBool, Ordering};
use std::sync::mpsc::{sync_channel, Receiver, RecvTimeoutError, SyncSender};
use std::sync::Arc;
use std::thread::{self, JoinHandle};
use std::time::Duration;
use wasapi::{DeviceEnumerator, Direction, SampleType, StreamMode, WaveFormat};
// 48 kHz stereo 32-bit float: 2 channels * 4 bytes = 8 bytes per frame.
const BLOCK_ALIGN: usize = 2 * 4;
pub struct WasapiLoopbackCapturer {
chunks: Receiver<Vec<f32>>,
channels: u32,
stop: Arc<AtomicBool>,
join: Option<JoinHandle<()>>,
}
impl WasapiLoopbackCapturer {
pub fn open(channels: u32) -> Result<WasapiLoopbackCapturer> {
anyhow::ensure!(
channels == 2,
"WASAPI loopback backend is stereo-only (got {channels})"
);
let (tx, rx) = sync_channel::<Vec<f32>>(64);
let stop = Arc::new(AtomicBool::new(false));
// Bring-up handshake: report open success/failure before returning, so a missing render
// endpoint surfaces as Err (caller continues without audio) rather than a silent dead thread.
let (ready_tx, ready_rx) = sync_channel::<Result<()>>(1);
let stop_t = stop.clone();
let join = thread::Builder::new()
.name("punktfunk-wasapi-audio".into())
.spawn(move || {
if let Err(e) = capture_thread(tx, stop_t, ready_tx) {
tracing::error!(error = format!("{e:#}"), "wasapi loopback thread failed");
}
})
.context("spawn wasapi audio thread")?;
match ready_rx.recv_timeout(Duration::from_secs(3)) {
Ok(Ok(())) => {
tracing::info!("WASAPI loopback capture: 48 kHz stereo f32 (default render endpoint)");
Ok(WasapiLoopbackCapturer {
chunks: rx,
channels,
stop,
join: Some(join),
})
}
Ok(Err(e)) => Err(e),
Err(_) => Err(anyhow!(
"wasapi loopback init timed out (no default render endpoint?)"
)),
}
}
}
impl Drop for WasapiLoopbackCapturer {
fn drop(&mut self) {
self.stop.store(true, Ordering::SeqCst);
if let Some(j) = self.join.take() {
let _ = j.join();
}
}
}
impl AudioCapturer for WasapiLoopbackCapturer {
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 WASAPI audio within 5s")),
Err(RecvTimeoutError::Disconnected) => Err(anyhow!("wasapi audio thread ended")),
}
}
fn channels(&self) -> u32 {
self.channels
}
fn drain(&mut self) {
while self.chunks.try_recv().is_ok() {}
}
}
fn capture_thread(
tx: SyncSender<Vec<f32>>,
stop: Arc<AtomicBool>,
ready: SyncSender<Result<()>>,
) -> Result<()> {
// COM must be initialized on THIS thread (MTA), before any device call.
if let Err(e) = wasapi::initialize_mta().ok().context("CoInitializeEx (MTA)") {
let _ = ready.send(Err(e));
return Ok(());
}
let res = (|| -> Result<()> {
// Loopback = capture the RENDER endpoint: get the default render device, but open a CAPTURE
// client with loopback=true over it.
let device = DeviceEnumerator::new()
.context("DeviceEnumerator")?
.get_default_device(&Direction::Render)
.context("default render endpoint (loopback needs a render device)")?;
let mut audio_client = device.get_iaudioclient().context("IAudioClient")?;
// 48 kHz stereo f32 interleaved; autoconvert lets WASAPI's shared-mode SRC match the engine
// mix format to ours, so we never resample in Rust. Loopback is implied by capturing a
// RENDER device with Direction::Capture in shared mode (wasapi sets STREAMFLAGS_LOOPBACK).
let desired = WaveFormat::new(32, 32, &SampleType::Float, SAMPLE_RATE as usize, 2, None);
let (default_period, _min_period) =
audio_client.get_device_period().context("device period")?;
let mode = StreamMode::EventsShared {
autoconvert: true,
buffer_duration_hns: default_period,
};
audio_client
.initialize_client(&desired, &Direction::Capture, &mode)
.context("initialize loopback client")?;
let h_event = audio_client.set_get_eventhandle().context("event handle")?;
let capture_client = audio_client
.get_audiocaptureclient()
.context("IAudioCaptureClient")?;
audio_client.start_stream().context("start loopback stream")?;
let _ = ready.send(Ok(()));
let mut bytes: VecDeque<u8> = VecDeque::new();
while !stop.load(Ordering::Relaxed) {
// Loopback fires events only while audio renders; the finite timeout keeps `stop` responsive.
if h_event.wait_for_event(100).is_err() {
continue;
}
loop {
match capture_client.get_next_packet_size() {
Ok(Some(0)) | Ok(None) => break,
Ok(Some(_n)) => {
capture_client
.read_from_device_to_deque(&mut bytes)
.context("read loopback")?;
}
Err(e) => return Err(anyhow!("get_next_packet_size: {e}")),
}
}
let whole = (bytes.len() / BLOCK_ALIGN) * BLOCK_ALIGN;
if whole == 0 {
continue;
}
let raw: Vec<u8> = bytes.drain(..whole).collect();
let mut samples = Vec::with_capacity(whole / 4);
for c in raw.chunks_exact(4) {
samples.push(f32::from_le_bytes([c[0], c[1], c[2], c[3]]));
}
let _ = tx.try_send(samples); // non-blocking, lossy — same discipline as PipeWire
}
audio_client.stop_stream().ok();
Ok(())
})();
if let Err(ref e) = res {
let _ = ready.send(Err(anyhow!("{e:#}")));
}
res
}
#[cfg(test)]
mod tests {
use super::*;
/// Live loopback round trip — skipped unless `PUNKTFUNK_WASAPI_LIVE=1` and a render endpoint
/// exists. Opens the capturer and pulls one chunk of interleaved f32.
#[test]
fn live_open_and_read() {
if std::env::var("PUNKTFUNK_WASAPI_LIVE").is_err() {
return;
}
let mut cap = match WasapiLoopbackCapturer::open(2) {
Ok(c) => c,
Err(e) => {
eprintln!("no render endpoint on this box ({e:#}) — skipping");
return;
}
};
assert_eq!(cap.channels(), 2);
match cap.next_chunk() {
Ok(samples) => assert!(samples.len() % 2 == 0, "interleaved stereo => even sample count"),
Err(e) => eprintln!("no audio within timeout (silent system?): {e:#}"),
}
}
}