86 commits of main, including the whole M1-M12 haptics sweep. Twelve conflicting files; three of them were more than textual. **The capability bits collided.** Both branches allocated the SAME wire bits for DIFFERENT features: `client_caps 0x04` and `host_caps 0x20` are redundant desktop audio on main and pad audio here. Merged naively, a peer would negotiate one and get the other. Pad audio moves to the next free bits — `CLIENT_CAP_PAD_AUDIO = 0x08`, `HOST_CAP_PAD_AUDIO = 0x40` — and the `abi.rs` mirrors move with them (their compile-time equality assertions caught the mismatch, which is exactly what they are for). **Both branches also claimed ABI v15.** Main's shipped (the rumble-policy floor), so the pad-audio surface becomes **v16**. **`native/input.rs` would have reintroduced a fixed bug.** This branch resets `rumble_seq[idx]` on pad removal; M1 established that the client's reorder gate is per-connection with no reset path, so restarting the host counter strands every later envelope until it climbs back. Took main's seq-preserving `clear_pad_feedback` and kept only the branch's `pad_streams.stop(idx)`. The rest: `wiring_plan::plan` now delegates to main's `plan_with_formats`, so the pad-endpoint filter moved into that body and the predicate behind it is factored out as `is_pad_render` (also what B10 needs); `Ds5Feedback::AUDIO` derives from main's `REPORT_ID_LEN` like its siblings; `AudioCtl` joins the explicitly-listed unhandled variants so the guard-false case is covered rather than swept up by a `_`; `include/punktfunk_core.h` regenerated rather than hand-merged.
210 lines
11 KiB
Rust
210 lines
11 KiB
Rust
//! Desktop audio capture for the GameStream audio stream. On Linux: a PipeWire stream that by
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//! default registers a host-owned **stream sink** claimed as the session's default output
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//! (apps play into it directly — immune to hardware-sink churn; `PUNKTFUNK_STREAM_SINK=0`
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//! falls back to recording the default sink's monitor). Either way the capture is delivered
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//! as interleaved `f32` PCM at 48 kHz in the requested channel count (stereo, 5.1 or 7.1 —
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//! GameStream surround order FL FR FC LFE RL RR [SL SR]). The audio data plane
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//! (`gamestream::audio`) reframes this into fixed Opus frames, encodes, and sends it.
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use anyhow::Result;
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/// Opus/GameStream audio is 48 kHz.
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pub const SAMPLE_RATE: u32 = 48_000;
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/// Stereo channel count — the default and the punktfunk/1 audio plane's fixed layout.
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pub const CHANNELS: usize = 2;
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/// Produces interleaved `f32` PCM at [`SAMPLE_RATE`] in the channel count it was opened
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/// with. Lives on its own thread; never blocks the capture loop (drops if the consumer
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/// falls behind).
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pub trait AudioCapturer: Send {
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/// Block until the next chunk of interleaved samples is available (variable size). The
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/// caller reframes into fixed Opus frames. An **empty** chunk means "no samples right now"
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/// (e.g. a quiet sink that hit the internal idle timeout) — NOT an error: the caller keeps the
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/// capturer. `Err` is reserved for a genuinely dead capture thread, signalling the caller to
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/// reopen.
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fn next_chunk(&mut self) -> Result<Vec<f32>>;
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/// The interleaved channel count this capturer delivers (what it was opened with).
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fn channels(&self) -> u32 {
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CHANNELS as u32
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}
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/// Discard any buffered chunks (called when a persistent capturer is reused for a new
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/// stream, so the client doesn't hear stale audio captured while idle). On Linux this is
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/// also the session-start hook: the stream-sink capturer re-claims the default sink here
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/// (see [`idle`](Self::idle)). Default: no-op.
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fn drain(&mut self) {}
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/// Called when a session parks the capturer back into the persistent slot: release
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/// session-scoped routing side effects while keeping the capture backend alive. On Linux
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/// the stream-sink capturer restores the user's default sink here, so host apps play to
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/// the real output again between streams; the claim returns with the next
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/// [`drain`](Self::drain) (reuse) or a fresh open. Default: no-op.
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fn idle(&mut self) {}
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}
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/// Open a live capturer for system output via PipeWire, asking for `channels` interleaved
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/// channels. Default: a host-owned stream sink claimed as the default output (the sink
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/// advertises exactly `channels`, so apps can produce real surround); with
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/// `PUNKTFUNK_STREAM_SINK=0`, the default sink's monitor, where a sink with fewer channels
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/// gets the missing positions filled with silence (zero upmix).
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#[cfg(target_os = "linux")]
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pub fn open_audio_capture(channels: u32) -> Result<Box<dyn AudioCapturer>> {
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linux::PwAudioCapturer::open(channels).map(|c| Box::new(c) as Box<dyn AudioCapturer>)
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}
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#[cfg(target_os = "windows")]
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pub fn open_audio_capture(channels: u32) -> Result<Box<dyn AudioCapturer>> {
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// The capture thread runs the audio wiring plan itself (audio_control::wire_now) before
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// resolving its endpoint — a fresh plan per open, because Windows endpoints churn — and
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// parks the default playback device on the plan's loopback endpoint (a silent sink by
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// default: audio plays on the client only) until the capturer is dropped.
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wasapi_cap::WasapiLoopbackCapturer::open(channels)
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.map(|c| Box::new(c) as Box<dyn AudioCapturer>)
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}
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#[cfg(not(any(target_os = "linux", target_os = "windows")))]
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pub fn open_audio_capture(_channels: u32) -> Result<Box<dyn AudioCapturer>> {
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anyhow::bail!("audio capture requires Linux + PipeWire or Windows + WASAPI")
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}
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/// Park a capturer at session end. Linux: store it in the persistent slot so the next session
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/// reuses it (no PipeWire thread churn). Windows: DROP it instead — closing the capture restores
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/// the operator's default playback device (it was parked on the loopback sink for the stream's
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/// lifetime, silencing the host), and a WASAPI reopen at the next session start is cheap and
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/// re-runs the wiring plan against the then-current endpoints.
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pub fn park_audio_capture(
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slot: &std::sync::Mutex<Option<Box<dyn AudioCapturer>>>,
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cap: Box<dyn AudioCapturer>,
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) {
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if cfg!(target_os = "windows") {
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drop(cap);
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} else {
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*slot.lock().unwrap() = Some(cap);
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}
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}
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/// The inverse of [`AudioCapturer`]: a virtual microphone the host *produces*. It registers a
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/// PipeWire `Audio/Source` node that host apps can record from; the host [`push`](Self::push)es
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/// decoded client-mic PCM (interleaved `f32` at [`SAMPLE_RATE`]) into it, and PipeWire delivers
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/// it to whichever app records the source — silence when no input is flowing. This is how the
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/// client's microphone reaches host applications (mic passthrough).
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///
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/// **Liveness contract.** Both backends run a worker thread that CAN die under the host's feet
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/// (Linux: the PipeWire daemon restarts with the session; Windows: the audio endpoint is
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/// invalidated/removed). A dead backend must be observable — [`push`](Self::push) returns `false`
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/// and [`alive`](Self::alive) turns false — so the owning [`MicPump`] drops the instance and
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/// reopens. Before this contract existed, a single backend death left `push` feeding a dead
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/// queue for the rest of the host's life: the historical "mic passthrough works on no host" bug.
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pub trait VirtualMic: Send {
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/// Push one chunk of interleaved `f32` PCM. Non-blocking — drops if the backend is behind
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/// (mic audio is lossy/real-time; a stale chunk is worse than a dropped one). Returns
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/// `false` iff the backend is DEAD (worker thread gone) — the caller must reopen; a merely
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/// congested backend drops the chunk and returns `true`.
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fn push(&self, pcm: &[f32]) -> bool;
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/// Backend liveness without pushing data — lets an idle pump notice a death between
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/// sessions, so the mic is already healthy again when the next client connects.
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fn alive(&self) -> bool;
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/// Drop any buffered-but-unplayed audio. Called after an uplink gap (client muted,
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/// session ended) so a recorder never hears a stale burst when audio resumes.
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fn discard(&self);
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/// The interleaved channel count the source was opened with.
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fn channels(&self) -> u32 {
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CHANNELS as u32
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}
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/// The adaptive de-jitter target (per-channel samples) the pump measured from uplink
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/// arrival jitter (see `mic_jitter`). A backend with a jitter ring primes around this PLUS
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/// one of its own consumer quanta: the ring must absorb arrival burstiness (the pump's
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/// number) and pull granularity (the backend's own), and neither may buy the other's depth
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/// — a 2048-frame recorder gets its one quantum, not three. Never called ⇒ the backend
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/// keeps its legacy fixed constants, which is also how `PUNKTFUNK_MIC_LEGACY_BUFFER=1`
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/// works (the pump simply never drives the target). Default: no ring, ignored.
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fn set_target_depth(&self, _samples_per_ch: usize) {}
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/// `(buffered, prime_target)` of the backend's jitter ring in per-channel samples — read
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/// by the pump's creep trim + telemetry. `None` while unknown (consumer not yet running,
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/// or a backend without a ring).
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fn depth(&self) -> Option<(usize, usize)> {
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None
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}
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/// Reset-on-read telemetry counters (see [`MicBackendStats`]). Default: all zero.
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fn take_stats(&self) -> MicBackendStats {
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MicBackendStats::default()
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}
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}
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/// Reset-on-read counters a [`VirtualMic`] backend reports into the pump's periodic mic
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/// telemetry line ("mic uplink health").
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#[derive(Debug, Default, Clone, Copy)]
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pub struct MicBackendStats {
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/// Full-drain re-prime arms: the ring emptied and gates on silence until the target depth
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/// rebuilds. One per talk spurt is normal; several per second mid-speech is the crackle.
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pub reprimes: u64,
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/// Per-channel samples dropped by the ring's overflow cap (drop-oldest).
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pub overflow_dropped: u64,
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}
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/// One-release escape hatch (docs: configuration → Audio / microphone):
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/// `PUNKTFUNK_MIC_LEGACY_BUFFER=1` keeps the pre-adaptive fixed mic buffering — the pump never
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/// drives the backend target (so the rings stay on their legacy constants: 48 ms prime /
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/// 120 ms cap on Windows, the 3-quanta clamp on Linux) and never creep-trims depth.
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pub(crate) fn mic_legacy_buffer() -> bool {
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static ON: std::sync::OnceLock<bool> = std::sync::OnceLock::new();
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*ON.get_or_init(|| std::env::var_os("PUNKTFUNK_MIC_LEGACY_BUFFER").is_some_and(|v| v != "0"))
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}
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/// Open a virtual microphone with `channels` interleaved channels (1 or 2). Linux: a PipeWire
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/// `Audio/Source`. Windows: writes into an existing virtual audio device's render endpoint (whose
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/// capture endpoint apps see as a mic) — see [`wasapi_mic`].
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#[cfg(target_os = "linux")]
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pub fn open_virtual_mic(channels: u32) -> Result<Box<dyn VirtualMic>> {
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linux::PwMicSource::open(channels).map(|m| Box::new(m) as Box<dyn VirtualMic>)
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}
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#[cfg(target_os = "windows")]
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pub fn open_virtual_mic(channels: u32) -> Result<Box<dyn VirtualMic>> {
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// The render thread runs the wiring plan itself (audio_control::wire_now) to resolve — and,
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// via the plan's default-device changes, to RESERVE — its target endpoint.
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wasapi_mic::WasapiVirtualMic::open(channels).map(|m| Box::new(m) as Box<dyn VirtualMic>)
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}
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#[cfg(not(any(target_os = "linux", target_os = "windows")))]
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pub fn open_virtual_mic(_channels: u32) -> Result<Box<dyn VirtualMic>> {
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anyhow::bail!("virtual mic requires Linux + PipeWire or Windows + a virtual audio device")
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}
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#[cfg(target_os = "windows")]
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#[path = "audio/windows/audio_control.rs"]
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mod audio_control;
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#[cfg(target_os = "linux")]
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mod linux;
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// DualSense pad-audio endpoint provisioning + loopback capture (design: pad haptics/audio).
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// pub(crate): the session layer queries endpoints by pad index and the CLI exposes the
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// `pad-endpoint` devtest.
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#[cfg(target_os = "windows")]
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#[path = "audio/windows/pad_endpoint.rs"]
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pub(crate) mod pad_endpoint;
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#[cfg(target_os = "windows")]
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#[path = "audio/windows/wasapi_cap.rs"]
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mod wasapi_cap;
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#[cfg(target_os = "windows")]
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#[path = "audio/windows/wasapi_mic.rs"]
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mod wasapi_mic;
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#[cfg_attr(not(target_os = "windows"), allow(dead_code))]
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#[path = "audio/wiring_plan.rs"]
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pub(crate) mod wiring_plan;
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// Pure capture-loop policy, split out for the same reason `wiring_plan` is: it encodes field
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// behaviour, so its tests must run on every platform's CI, not only Windows.
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#[cfg_attr(not(target_os = "windows"), allow(dead_code))]
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#[path = "audio/capture_policy.rs"]
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pub(crate) mod capture_policy;
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mod mic_jitter;
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mod mic_pump;
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pub use mic_pump::{MicFrame, MicPump};
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