Fix audio clock drift and adaptive jitter buffering

docs/api-dotnet.md

@@ -177,8 +177,10 @@ Noise reduction does not gate speech.
 `TimeProvider` timestamps; tests inject a clock. Threshold changes are atomic; all
 processing state otherwise has one owner. Codec/DSP processing methods allocate no
 managed memory after initialization, verified across 1,000 combined cycles. They run
-on a managed worker, never the native real-time device callback. Native device rings,
-jitter, mixer, and audio scheduling remain later work.
+on a managed worker, never the native real-time device callback. Managed capture/playback
+rings compensate independent clock drift around a configurable 20/40/60 ms target. Receive
+jitter is duration-aware, reserves one codec frame for DRED/FEC look-ahead, and decodes each
+channel packet at its actual duration before the fixed 20 ms mixer stage.
 
 ## Initial managed server
 

docs/voice.md

@@ -117,6 +117,9 @@ Guidance baked into defaults / docs:
   (accumulating two 960-frames for a 40 ms channel, splitting each into two 480-frames for a
   10 ms channel, etc.). This keeps the hardware/`vc_stream_feed_pcm` contract a single 48 kHz
   clock regardless of the channel's window — see `vc_client::on_capture_frame`.
+  The device/mixer quantum is not the Opus packet duration: managed send streams reframe it
+  into the channel's 5/10/20/40/60 ms packets, and receive streams decode at that duration
+  before slicing decoded PCM back into 20 ms mixer blocks.
 - **Mode/bitrate:** speech channels → `MONO`, `VOIP`, 24–32 kbps, DTX on, FEC on.
   Music/screen-audio channels → `STEREO`, `AUDIO`, 96–128 kbps, DTX off, FEC optional.
 - **`application`:** `VOIP` for talk, `AUDIO` for music/screen-share, `LOWDELAY` for
@@ -152,26 +155,30 @@ straight through to PLC.
 ## 5. Jitter buffer
 
 Each receiver keeps an **adaptive jitter buffer per ssrc** with **bounded-depth playout**
-(`core/src/audio/audio_engine.cpp` — `JitterBuffer` + `AudioEngine::on_playback`).
+(`dotnet/src/VoiceCat.Audio/ReceiveStream.cs`).
 
 - Frames are inserted by `timestamp`; playback reads in order at the device callback rate.
 - **The playout clock is always bounded against the stream's *leading edge* (newest buffered
-  frame), never re-synced to the oldest.** The clock free-runs at the playback hardware rate,
+  frame), never re-synced to the oldest.** The managed playout clock is isolated from hardware
+  drift by the adaptive PCM ring,
   while the sender omits VAD/PTT/DTX silence from its timestamps, so the two diverge across gaps
   and late joins. Two corrections keep latency bounded:
   - **(Re)seed to the leading edge** on first frame, on a talkspurt `marker`, or when the clock
-    has run past the newest frame (starved after silence). No artificial prebuffer — latency
-    starts as low as possible; buffered frames still play oldest-first.
+    has run past the newest frame (starved after silence). Playout retains the adaptive target;
+    when DRED/FEC is enabled that includes one codec frame of recovery look-ahead.
   - **Frame-skip catch-up:** when the backlog grows past `target + hysteresis` (clock drift,
     bursty arrival, reordering), fast-forward the clock to leave `target` buffered and drop the
     now-stale frames. This is the downward force that prevents latency from ratcheting upward.
-- `target` is the adaptive jitter estimate (EWMA of inter-arrival gap vs. the per-frame gap),
-  floored; silence gaps and reordered stragglers are rejected as outliers so they don't inflate
-  it. The late-drop window tracks `target` (floored/capped at 500 ms).
+- `target` is an EWMA of arrival-gap variation measured in 48 kHz sample time. DRED or FEC
+  reserves one complete channel Opus frame of look-ahead, variation can raise the target to
+  120 ms, and packet history is bounded to 500 ms. Limits are durations rather than packet
+  counts, so 5 ms and 60 ms channels receive the same policy.
 - Late frames past the playout point are dropped; gaps are filled by DRED (if the next frame
   arrived) or PLC.
 - The `marker` flag (start of talkspurt) — set by the sender on the first frame after a
   transmission gap — lets the buffer reseed cleanly after silence/DTX without accumulating drift.
+- One late device callback does not end a talkspurt. Capture resets only after 200 ms of
+  continuous starvation, avoiding a marker/rebuffer cascade from an isolated scheduling miss.
 - Diagnostics per stream: `packets_lost`, `duplicates`, `underruns`, `target_depth_ms`.
 
 ```
@@ -180,6 +187,11 @@ Each receiver keeps an **adaptive jitter buffer per ssrc** with **bounded-depth
                              jitter estimate feeds depth
 ```
 
+Capture and playback use an allocation-free adaptive PCM ring between the managed 20 ms clock
+and the hardware clock. Linear-interpolation correction, limited to ±0.5%, holds the ring near
+its target instead of periodically dropping a block or rendering silence as device clocks drift.
+The local **Audio buffering** preset is 20, 40 (default), or 60 ms and is persisted per client.
+
 ## 6. UDP keepalive & NAT
 
 - A `KEEPALIVE` (type 2) frame flows both directions on the media channel every ~5 s to