fix(host/windows): don't 2-way-split-encode Main10 — it's SLOWER on Ada (fixes broken HDR animations)

The "broken animations in HDR" was an encode-throughput cliff, not the ACCESS_LOST churn. Measured at
5120x1440@240 HEVC Main10 on the RTX 4090: forced 2-way split-encode = 7.6 ms/frame (~131 fps, well
over the 4.17 ms/240fps budget → choppy), while SINGLE engine = 2.8-3.9 ms/frame (~256-357 fps, fits
240). The split/merge overhead dominates for 10-bit; a single Ada NVENC engine already handles 5K@240
Main10 comfortably. So the split decision now forces DISABLE for Main10 (bit_depth >= 10), keeping the
existing forced-2 only for 8-bit above 1 Gpix/s. PUNKTFUNK_SPLIT_ENCODE still overrides. Added a
split-mode log line.

Validated live on the 4090: encode_us_p50 7.6 ms → 3.9 ms at 5K240 HDR with no env override.

Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>

crates/punktfunk-client-windows/src/audio.rs

@@ -0,0 +1,285 @@
+//! Audio: playback (decoded PCM → a WASAPI shared-mode render stream) and the microphone
+//! uplink (WASAPI capture → Opus → 0xCB datagrams, the inverse of the host's virtual mic).
+//!
+//! The WASAPI analogue of the Linux client's PipeWire backend. Playback mirrors the host's
+//! virtual-mic producer's adaptive jitter buffer: the session pump pushes 5 ms Opus-decoded
+//! chunks on the network clock; the WASAPI render thread pulls whole event-driven quanta on
+//! the device clock. Prime to ~3 quanta before producing, cap the ring so latency stays
+//! bounded, re-prime after a real drain.
+//!
+//! WASAPI objects are COM-apartment-bound and not `Send`, so they live on a dedicated thread
+//! (the same discipline as the host's `wasapi_cap`); only the channel + stop flag + join
+//! handle cross the boundary.
+
+use anyhow::{anyhow, Context, Result};
+use punktfunk_core::client::NativeClient;
+use std::collections::VecDeque;
+use std::sync::atomic::{AtomicBool, Ordering};
+use std::sync::mpsc::{Receiver, SyncSender, TrySendError};
+use std::sync::Arc;
+use std::time::Duration;
+use wasapi::{Direction, SampleType, StreamMode, WaveFormat};
+
+const SAMPLE_RATE: usize = 48_000;
+const CHANNELS: usize = 2;
+/// 48 kHz stereo f32: 2 channels * 4 bytes = 8 bytes per frame.
+const BLOCK_ALIGN: usize = CHANNELS * 4;
+/// Mic frames are 20 ms (960 samples/channel) — any size ≤ 120 ms is fine host-side.
+const MIC_FRAME: usize = 960;
+
+pub struct AudioPlayer {
+    pcm_tx: SyncSender<Vec<f32>>,
+    stop: Arc<AtomicBool>,
+    thread: Option<std::thread::JoinHandle<()>>,
+}
+
+impl AudioPlayer {
+    /// Spawn the WASAPI render thread. Failure (no render endpoint on this box) is
+    /// survivable — the caller streams video-only.
+    pub fn spawn() -> Result<AudioPlayer> {
+        // 64 × 5 ms = 320 ms of slack between the pump and the WASAPI loop.
+        let (pcm_tx, pcm_rx) = std::sync::mpsc::sync_channel::<Vec<f32>>(64);
+        let stop = Arc::new(AtomicBool::new(false));
+        let (ready_tx, ready_rx) = std::sync::mpsc::sync_channel::<Result<()>>(1);
+        let stop_t = stop.clone();
+        let thread = std::thread::Builder::new()
+            .name("punktfunk-audio".into())
+            .spawn(move || {
+                if let Err(e) = render_thread(pcm_rx, stop_t, ready_tx) {
+                    tracing::warn!(error = format!("{e:#}"), "audio playback thread ended");
+                }
+            })
+            .context("spawn audio thread")?;
+        match ready_rx.recv_timeout(Duration::from_secs(3)) {
+            Ok(Ok(())) => {
+                tracing::info!("WASAPI render: 48 kHz stereo f32 (default endpoint)");
+                Ok(AudioPlayer {
+                    pcm_tx,
+                    stop,
+                    thread: Some(thread),
+                })
+            }
+            Ok(Err(e)) => Err(e),
+            Err(_) => Err(anyhow!(
+                "wasapi render init timed out (no render endpoint?)"
+            )),
+        }
+    }
+
+    /// Queue one interleaved-stereo f32 chunk. Drops the chunk if the WASAPI 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) {
+        self.stop.store(true, Ordering::SeqCst);
+        if let Some(t) = self.thread.take() {
+            let _ = t.join();
+        }
+    }
+}
+
+fn render_thread(
+    pcm_rx: Receiver<Vec<f32>>,
+    stop: Arc<AtomicBool>,
+    ready: SyncSender<Result<()>>,
+) -> Result<()> {
+    if let Err(e) = wasapi::initialize_mta()
+        .ok()
+        .context("CoInitializeEx (MTA)")
+    {
+        let _ = ready.send(Err(e));
+        return Ok(());
+    }
+    let res = (|| -> Result<()> {
+        let device =
+            wasapi::get_default_device(&Direction::Render).context("default render endpoint")?;
+        let mut audio_client = device.get_iaudioclient().context("IAudioClient")?;
+        let desired = WaveFormat::new(32, 32, &SampleType::Float, SAMPLE_RATE, CHANNELS, 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::Render, &mode)
+            .context("initialize render client")?;
+        let h_event = audio_client.set_get_eventhandle().context("event handle")?;
+        let render_client = audio_client
+            .get_audiorenderclient()
+            .context("IAudioRenderClient")?;
+        audio_client.start_stream().context("start render stream")?;
+        let _ = ready.send(Ok(()));
+
+        // Adaptive jitter buffer, in f32-byte units (same shape as the host's virtual mic).
+        let mut ring: VecDeque<u8> = VecDeque::new();
+        let mut primed = false;
+
+        while !stop.load(Ordering::Relaxed) {
+            if h_event.wait_for_event(100).is_err() {
+                continue;
+            }
+            // Drain everything the pump has queued into the ring.
+            while let Ok(chunk) = pcm_rx.try_recv() {
+                for s in chunk {
+                    ring.extend(s.to_le_bytes());
+                }
+            }
+            let avail_frames = audio_client
+                .get_available_space_in_frames()
+                .context("available space")? as usize;
+            if avail_frames == 0 {
+                continue;
+            }
+            let want_bytes = avail_frames * BLOCK_ALIGN;
+
+            // Prime to ~3 quanta; cap at ~1 quantum of slack beyond that; re-prime on drain.
+            let target = (3 * want_bytes).clamp(720 * BLOCK_ALIGN, 9600 * BLOCK_ALIGN);
+            while ring.len() > target.max(want_bytes) + want_bytes {
+                ring.pop_front();
+            }
+            if !primed && ring.len() >= target {
+                primed = true;
+            }
+
+            let mut out = vec![0u8; want_bytes];
+            if primed {
+                let n = ring.len().min(want_bytes);
+                for (dst, b) in out.iter_mut().zip(ring.drain(..n)) {
+                    *dst = b;
+                }
+            }
+            if ring.is_empty() {
+                primed = false;
+            }
+            render_client
+                .write_to_device(avail_frames, BLOCK_ALIGN, &out, None)
+                .context("write_to_device")?;
+        }
+        audio_client.stop_stream().ok();
+        Ok(())
+    })();
+    if let Err(ref e) = res {
+        let _ = ready.send(Err(anyhow!("{e:#}")));
+    }
+    res
+}
+
+/// The microphone uplink: capture the default input device, Opus-encode 20 ms chunks, ship
+/// them as 0xCB datagrams into the host's virtual mic source.
+pub struct MicStreamer {
+    stop: Arc<AtomicBool>,
+    thread: Option<std::thread::JoinHandle<()>>,
+}
+
+impl MicStreamer {
+    pub fn spawn(connector: Arc<NativeClient>) -> Result<MicStreamer> {
+        let stop = Arc::new(AtomicBool::new(false));
+        let stop_t = stop.clone();
+        let thread = std::thread::Builder::new()
+            .name("punktfunk-mic".into())
+            .spawn(move || {
+                if let Err(e) = mic_thread(&connector, stop_t) {
+                    tracing::warn!(error = format!("{e:#}"), "mic uplink thread ended");
+                }
+            })
+            .context("spawn mic thread")?;
+        Ok(MicStreamer {
+            stop,
+            thread: Some(thread),
+        })
+    }
+}
+
+impl Drop for MicStreamer {
+    fn drop(&mut self) {
+        self.stop.store(true, Ordering::SeqCst);
+        if let Some(t) = self.thread.take() {
+            let _ = t.join();
+        }
+    }
+}
+
+fn mic_thread(connector: &Arc<NativeClient>, stop: Arc<AtomicBool>) -> Result<()> {
+    wasapi::initialize_mta()
+        .ok()
+        .context("CoInitializeEx (MTA)")?;
+
+    let mut encoder = opus::Encoder::new(
+        SAMPLE_RATE as u32,
+        opus::Channels::Stereo,
+        opus::Application::Voip,
+    )
+    .map_err(|e| anyhow!("opus encoder: {e}"))?;
+    let _ = encoder.set_bitrate(opus::Bitrate::Bits(64_000));
+
+    let device = wasapi::get_default_device(&Direction::Capture)
+        .context("default capture endpoint (no microphone?)")?;
+    let mut audio_client = device.get_iaudioclient().context("IAudioClient")?;
+    let desired = WaveFormat::new(32, 32, &SampleType::Float, SAMPLE_RATE, CHANNELS, 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 capture 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 capture stream")?;
+
+    let mut bytes: VecDeque<u8> = VecDeque::new();
+    let mut ring: VecDeque<f32> = VecDeque::new();
+    let mut out = vec![0u8; 4000];
+    let mut seq = 0u32;
+
+    while !stop.load(Ordering::Relaxed) {
+        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 capture")?;
+                }
+                Err(e) => return Err(anyhow!("get_next_packet_size: {e}")),
+            }
+        }
+        let whole = (bytes.len() / 4) * 4;
+        for c in bytes.drain(..whole).collect::<Vec<u8>>().chunks_exact(4) {
+            ring.push_back(f32::from_le_bytes([c[0], c[1], c[2], c[3]]));
+        }
+        // Ship every complete 20 ms stereo frame.
+        while ring.len() >= MIC_FRAME * CHANNELS {
+            let pcm: Vec<f32> = ring.drain(..MIC_FRAME * CHANNELS).collect();
+            match encoder.encode_float(&pcm, &mut 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 _ = connector.send_mic(seq, pts, out[..len].to_vec());
+                    seq = seq.wrapping_add(1);
+                }
+                Err(e) => tracing::debug!(error = %e, "opus mic encode"),
+            }
+        }
+    }
+    audio_client.stop_stream().ok();
+    Ok(())
+}