refactor(state_portal): extract bitrate helpers + thread loops to submodules

Step 3: split state_portal.rs (1241 -> 829 LOC) into three modules.

- src/state_portal.rs (829 LOC): keeps StatePortal struct + impl (with
  poll_and_encode / handle_pw_frame / shutdown / etc.) + Drop + PortalStage
  enum + DRM helpers + DRM tests. Per Oracle/Explore audit, all 21
  StatePortal fields are private and poll_and_encode interleaves three
  channel reads with state-machine transitions; moving it would force
  pub(crate) on every field, so it stays in mod.rs.
- src/state_portal/bitrate.rs (144 LOC): RESOLUTION_TIERS + 4 pure fns
  (resolution_bitrate_bps / webrtc_startup_bitrate_bps / select_resolution /
  next_upscale_tier) + 10 tests that exercise them. Pure fns with no
  StatePortal field access — the cleanest possible extract.
- src/state_portal/threads.rs (287 LOC): the 5 thread-related types
  (EncodeThreadTiming / EncodeThread / WebrtcThread / WebRtcThreadConfig /
  WebRtcThreadChannels) + the two free fns encode_thread_loop /
  webrtc_thread_loop + the 3 channel-semantics regression tests
  (try_send_* / shutdown_rx_drop_*) that document crossbeam invariants
  the shutdown logic relies on. Struct fields widened to pub(super) so
  StatePortal in mod.rs can construct and join them.

Test preservation:
- state_portal test count: 17 (mod.rs=4 drm tests + bitrate.rs=10 +
  threads.rs=3 channel tests) — matches baseline.

Verification (all green):
- cargo build / cargo build --release
- cargo test (79 lib + 3 integration = 82 pass, 1 ignored — unchanged)
- cargo clippy --all-targets -- -D warnings
- cargo fmt --check

src/state_portal/threads.rs

@@ -0,0 +1,287 @@
+use std::sync::atomic::{AtomicBool, Ordering};
+use std::sync::Arc;
+use std::time::{Duration, Instant};
+
+use crate::avhw::{BitrateCommand, CpuNv12Frame, EncodeOutcome, EncodedH264Frame, SwEncEncode};
+use crate::webrtc::WebRtcState;
+
+use super::bitrate::{next_upscale_tier, resolution_bitrate_bps, select_resolution};
+
+pub(super) struct EncodeThreadTiming {
+    pub(super) sws_us: u64,
+    pub(super) encode_us: u64,
+    pub(super) output_bytes: usize,
+}
+
+pub(super) struct EncodeThread {
+    pub(super) handle: Option<std::thread::JoinHandle<()>>,
+    pub(super) input_tx: crossbeam_channel::Sender<CpuNv12Frame>,
+    pub(super) timing_rx: crossbeam_channel::Receiver<EncodeThreadTiming>,
+    pub(super) duplicate_count: std::sync::Arc<std::sync::atomic::AtomicU64>,
+}
+
+pub(super) struct WebrtcThread {
+    pub(super) handle: Option<std::thread::JoinHandle<()>>,
+    pub(super) sent_gap_rx: crossbeam_channel::Receiver<(f64, Option<f64>)>,
+}
+
+/// Static configuration handed to the WebRTC sender thread. Immutable for the
+/// thread's lifetime; a resolution tier change rebuilds the whole pipeline
+/// (and spawns a new thread) rather than mutating this.
+pub(super) struct WebRtcThreadConfig {
+    pub(super) fps: u32,
+    pub(super) enc_width: u32,
+    pub(super) enc_height: u32,
+    pub(super) max_bitrate: u64,
+}
+
+/// Channel endpoints owned exclusively by the WebRTC sender thread after spawn.
+/// The reverse endpoints stay with StatePortal (or the encode thread) for
+/// inbound/outbound traffic.
+pub(super) struct WebRtcThreadChannels {
+    pub(super) webrtc_rx: crossbeam_channel::Receiver<EncodedH264Frame>,
+    pub(super) sent_gap_tx: crossbeam_channel::Sender<(f64, Option<f64>)>,
+    pub(super) bitrate_tx: crossbeam_channel::Sender<BitrateCommand>,
+    pub(super) resolution_tx: crossbeam_channel::Sender<BitrateCommand>,
+}
+
+pub(super) fn encode_thread_loop(
+    mut encode: SwEncEncode,
+    input_rx: crossbeam_channel::Receiver<CpuNv12Frame>,
+    timing_tx: crossbeam_channel::Sender<EncodeThreadTiming>,
+    duplicate_count: std::sync::Arc<std::sync::atomic::AtomicU64>,
+) {
+    loop {
+        match input_rx.recv() {
+            Ok(frame) => {
+                match encode.encode_cpu_frame(&frame) {
+                    Ok(EncodeOutcome::Encoded) => {
+                        let t = encode.take_timing();
+                        let _ = timing_tx.try_send(EncodeThreadTiming {
+                            sws_us: t.sws_us,
+                            encode_us: t.encode_us,
+                            output_bytes: t.output_bytes,
+                        });
+                    }
+                    Ok(EncodeOutcome::SkippedDuplicate) => {
+                        duplicate_count.fetch_add(1, std::sync::atomic::Ordering::Relaxed);
+                    }
+                    Ok(_) => {
+                        // SkippedPaused / SkippedDisconnected — no counter needed
+                    }
+                    Err(e) => {
+                        tracing::error!("Encode thread error: {e}");
+                        break;
+                    }
+                }
+            }
+            Err(_) => {
+                tracing::info!("Encode thread input closed, flushing encoder");
+                if let Err(e) = encode.flush() {
+                    tracing::error!("Encode thread flush error: {e}");
+                }
+                break;
+            }
+        }
+    }
+    tracing::info!("Encode thread exiting");
+}
+
+pub(super) fn webrtc_thread_loop(
+    mut wrtc: WebRtcState,
+    config: WebRtcThreadConfig,
+    channels: WebRtcThreadChannels,
+    paused: Arc<AtomicBool>,
+) {
+    let WebRtcThreadConfig {
+        fps,
+        enc_width,
+        enc_height,
+        max_bitrate,
+    } = config;
+    let WebRtcThreadChannels {
+        webrtc_rx,
+        sent_gap_tx,
+        bitrate_tx,
+        resolution_tx,
+    } = channels;
+    let mut frames_sent: u64 = 0;
+    let mut last_send: Option<std::time::Instant> = None;
+    let mut last_sent_bitrate: Option<u64> = None;
+    let initial_tier = (enc_width, enc_height);
+    let mut current_tier = initial_tier;
+    let mut upscale_counter = 0u32;
+    let mut last_resolution_eval = Instant::now();
+    let timeout = Duration::from_millis(1);
+
+    loop {
+        if let Err(e) = wrtc.handle_signaling() {
+            tracing::error!("WebRTC signaling error: {e}");
+            break;
+        }
+        if let Err(e) = wrtc.poll_and_feed() {
+            tracing::error!("WebRTC poll error: {e}");
+            break;
+        }
+
+        if wrtc.take_force_keyframe() {
+            let _ = bitrate_tx.try_send(BitrateCommand::ForceKeyframe);
+        }
+
+        let connected = wrtc.is_connected();
+        let was_paused = paused.load(Ordering::Relaxed);
+        let now_paused = !connected;
+        if was_paused && !now_paused {
+            tracing::info!("WebRTC client connected, resuming encoding");
+        } else if !was_paused && now_paused {
+            tracing::warn!("WebRTC client disconnected, pausing encoding");
+        }
+        paused.store(now_paused, Ordering::Relaxed);
+
+        if let Some(bwe) = wrtc.get_bwe_estimate() {
+            // #23: Cap BWE to prevent runaway bitrate escalation. Without this, BWE
+            // estimates can rise to 10+ Mbps, causing IDR bursts and PLI storms.
+            let effective_bwe = bwe.min(max_bitrate);
+            if effective_bwe != bwe {
+                tracing::debug!(
+                    bwe,
+                    effective_bwe,
+                    max_bitrate,
+                    "BWE exceeds --max-bitrate cap, clamping"
+                );
+            }
+            let bwe = effective_bwe;
+
+            let should_send = match last_sent_bitrate {
+                None => true,
+                Some(last) => {
+                    let diff = bwe.abs_diff(last);
+                    diff * 10 > last
+                }
+            };
+            if should_send {
+                let _ = bitrate_tx.try_send(BitrateCommand::UpdateBitrate { target_bps: bwe });
+                last_sent_bitrate = Some(bwe);
+            }
+
+            if last_resolution_eval.elapsed() >= Duration::from_secs(1) {
+                last_resolution_eval = Instant::now();
+                let selected = select_resolution(current_tier.0, current_tier.1, bwe, fps);
+                if selected != current_tier {
+                    current_tier = selected;
+                    upscale_counter = 0;
+                    let _ = resolution_tx.try_send(BitrateCommand::UpdateResolution {
+                        width: current_tier.0,
+                        height: current_tier.1,
+                    });
+                    wrtc.set_need_keyframe();
+                } else if let Some(next_tier) = next_upscale_tier(current_tier, initial_tier) {
+                    let needed = resolution_bitrate_bps(next_tier.0, next_tier.1, fps);
+                    if bwe > needed.saturating_mul(120) / 100 {
+                        upscale_counter = upscale_counter.saturating_add(1);
+                        if upscale_counter >= 10 {
+                            current_tier = next_tier;
+                            upscale_counter = 0;
+                            let _ = resolution_tx.try_send(BitrateCommand::UpdateResolution {
+                                width: current_tier.0,
+                                height: current_tier.1,
+                            });
+                            wrtc.set_need_keyframe();
+                        }
+                    } else {
+                        upscale_counter = 0;
+                    }
+                } else {
+                    upscale_counter = 0;
+                }
+            }
+        }
+
+        if connected {
+            while let Ok(enc_frame) = webrtc_rx.try_recv() {
+                if let Err(e) = wrtc.write_h264_frame(&enc_frame.data, enc_frame.pts_ticks) {
+                    tracing::debug!("WebRTC write frame error: {e}");
+                }
+                frames_sent = frames_sent.saturating_add(1);
+                let gap_ms = last_send
+                    .map(|l| l.elapsed().as_secs_f64() * 1000.0)
+                    .unwrap_or(0.0);
+                // Compute capture-to-send age on the sending thread so the
+                // frame_age stat stays accurate when batch-drained later.
+                let age_ms = Some(enc_frame.capture_time.elapsed().as_secs_f64() * 1000.0);
+                last_send = Some(std::time::Instant::now());
+                let _ = sent_gap_tx.try_send((gap_ms, age_ms));
+            }
+        } else {
+            while webrtc_rx.try_recv().is_ok() {}
+        }
+
+        match webrtc_rx.recv_timeout(timeout) {
+            Ok(enc_frame) => {
+                if wrtc.is_connected() {
+                    if let Err(e) = wrtc.write_h264_frame(&enc_frame.data, enc_frame.pts_ticks) {
+                        tracing::debug!("WebRTC write frame error: {e}");
+                    }
+                    frames_sent = frames_sent.saturating_add(1);
+                    let gap_ms = last_send
+                        .map(|l| l.elapsed().as_secs_f64() * 1000.0)
+                        .unwrap_or(0.0);
+                    let age_ms = Some(enc_frame.capture_time.elapsed().as_secs_f64() * 1000.0);
+                    last_send = Some(std::time::Instant::now());
+                    let _ = sent_gap_tx.try_send((gap_ms, age_ms));
+                }
+            }
+            Err(crossbeam_channel::RecvTimeoutError::Timeout) => {}
+            Err(crossbeam_channel::RecvTimeoutError::Disconnected) => {
+                tracing::info!("WebRTC channel disconnected, exiting thread");
+                return;
+            }
+        }
+    }
+
+    tracing::info!("WebRTC thread exiting");
+}
+
+#[cfg(test)]
+mod tests {
+
+    #[test]
+    fn try_send_full_channel_returns_full_not_block() {
+        let (tx, rx) = crossbeam_channel::bounded::<Vec<u8>>(2);
+        tx.send(vec![1]).unwrap();
+        tx.send(vec![2]).unwrap();
+
+        assert!(matches!(
+            tx.try_send(vec![3]),
+            Err(crossbeam_channel::TrySendError::Full(_))
+        ));
+        assert_eq!(rx.len(), 2);
+    }
+
+    #[test]
+    fn try_send_after_rx_dropped_returns_disconnected() {
+        let (tx, rx) = crossbeam_channel::bounded::<Vec<u8>>(2);
+        drop(rx);
+
+        assert!(matches!(
+            tx.try_send(vec![1]),
+            Err(crossbeam_channel::TrySendError::Disconnected(_))
+        ));
+    }
+
+    // given: full bounded channel
+    // when:  rx is dropped, then try_send
+    // expect: Disconnected, not blocking
+    #[test]
+    fn shutdown_rx_drop_prevents_deadlock_on_full_channel() {
+        let (tx, rx) = crossbeam_channel::bounded::<Vec<u8>>(2);
+        tx.send(vec![1]).unwrap();
+        tx.send(vec![2]).unwrap();
+        drop(rx);
+
+        assert!(matches!(
+            tx.try_send(vec![3]),
+            Err(crossbeam_channel::TrySendError::Disconnected(_))
+        ));
+    }
+}