[client] audio: harden low-latency streaming

Rework audio provider and backend lifecycles so playback and capture
callbacks quiesce without blocking real-time threads. Move activation,
teardown, controls, retries, and diagnostics onto bounded workers.

Harden USB audio cadence, feedback, and capture recovery.
Preserve source clocks through recording and pace packets from the
device clock. Bound queues, waits, conversion buffers, and packet sizes.

Make PipeWire and PulseAudio stream control thread-safe and recoverable.
Correct latency clock domains, coalesce rate updates, preserve recent
capture under overload, and keep logging outside real-time callbacks.

client/include/interface/audio.h

@@ -106,7 +106,8 @@ typedef struct LG_AudioEventOps
 {
   /* Format and clock pointers are borrowed for the duration of each call.
    * A NULL source clock indicates that the provider has no usable clock.
-   * Providers must serialize event delivery for an attachment. Stream
+   * Providers must serialize event delivery within each stream direction.
+   * Playback and recording events may be delivered concurrently. Stream
    * generations are nonzero and uniquely identify each stream instance. */
   void (*playbackStart)(void * opaque, uint32_t generation,
       const LG_AudioFormat * format, const LG_AudioClock * sourceClock);
@@ -163,7 +164,9 @@ typedef struct LG_AudioOps
    * audio callback with the measured playback device clock. Its position uses
    * the device's independent output-frame timeline and its time includes the
    * backend's estimated presentation latency. targetRate is the source frame
-   * rate requested by the client's buffer controller. */
+   * rate requested by the client's buffer controller. Return false while
+   * active rate feedback is unavailable; sustained failure makes the client
+   * restart playback with local rate control. */
   bool (*clockFeedback)(void * opaque, uint32_t generation,
       const LG_AudioClock * playbackClock, double targetRate);
 }