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>

crates/punktfunk-host/src/m3.rs

@@ -105,7 +105,7 @@ async fn serve(opts: M3Options) -> Result<()> {
     );
 
     // One audio capturer for the whole host lifetime, handed from session to session
-    // (PipeWire streams have no cheap teardown — see AudioCapSlot).
+    // (avoids a PipeWire stream setup per session — see AudioCapSlot).
     let audio_cap: AudioCapSlot = Arc::new(std::sync::Mutex::new(None));
 
     let mut served = 0u32;
@@ -143,8 +143,8 @@ async fn serve(opts: M3Options) -> Result<()> {
 const HANDSHAKE_TIMEOUT: std::time::Duration = std::time::Duration::from_secs(10);
 
 /// Persistent audio-capturer slot, reused across sessions (same pattern as the GameStream
-/// path): `PwAudioCapturer` has no teardown — dropping one per session would leak its
-/// PipeWire thread + core connection + live capture node on the daemon every session.
+/// path): keeps one warm PipeWire capture stream instead of a connect/negotiate cycle —
+/// and a daemon-side node churn — per session. (Drop now tears a capturer down cleanly.)
 type AudioCapSlot = Arc<std::sync::Mutex<Option<Box<dyn crate::audio::AudioCapturer>>>>;
 
 /// One client session: handshake → input/audio planes → data plane until done/disconnect.
@@ -459,7 +459,7 @@ fn audio_thread(conn: quinn::Connection, stop: Arc<AtomicBool>, audio_cap: Audio
             c.drain(); // discard audio captured between sessions
             c
         }
-        None => match crate::audio::open_audio_capture() {
+        None => match crate::audio::open_audio_capture(CHANNELS as u32) {
             Ok(c) => c,
             Err(e) => {
                 tracing::warn!(error = %format!("{e:#}"), "punktfunk/1 audio unavailable — session continues without it");