[client] audio: redesign playback synchronization

Model playback latency from the device period, arrival jitter, source
packet phase, and resampler delay. Treat audio:latencyOffset as an
explicit addition to this minimum and align the first device pull to
the next packet deadline. This starts playback near its steady-state
target without unnecessary prefill or startup underruns. Use a
512-frame default period as a practical low-latency baseline.

Replace the startup clock hold with a one-sided proportional
acquisition controller, then hand off to source/device rate
feed-forward and a slow phase loop. Calibrate the logical device
timeline at handoff, discard correction that opposes the current
error, and slew-limit rate changes. This prevents startup drain,
integral wind-up, overshoot, and long convergence while preserving
clock-drift compensation.

Allow audio backends to expose a real-time resampler and use
PipeWire's adaptive resampler when version 1.4 or newer supports it.
Retain libsamplerate as the fallback and add audio:resampler to select
the implementation. Wait for PipeWire stream setup to complete and
propagate rate-control failures cleanly.

Track PipeWire input-consumption and output-equivalent clocks
separately. The input clock measures ring latency while the output
clock drives feed-forward using the ratio that governed each request.
This removes delayed self-feedback that made adaptive resampling
oscillate between the correction limits.

Reduce audio:debug output to useful latency, clock, jitter, and xrun
values, and scale the playback graph from the startup estimate. Update
the option names and documentation for the new latency model.

client/include/interface/audiodev.h

@@ -45,10 +45,13 @@ struct LG_AudioDevOps
   struct
   {
     /* setup the stream for playback but don't start it yet, returning false
-     * if the stream could not be configured
+     * if the stream could not be configured. If backend resampling is
+     * requested, resamplerEnabled reports whether it was activated for this
+     * stream.
      * Note: the pull function returns f32 samples
      */
     bool (*setup)(int channels, int sampleRate, int requestedPeriodFrames,
+      bool requestResampler, bool * resamplerEnabled,
       int * maxPeriodFrames, int * startFrames, LG_AudioPullFn pullFn);
 
     /* called when there is data available to start playback */
@@ -63,6 +66,10 @@ struct LG_AudioDevOps
     /* [optional] called to set muting of the output */
     void (*mute)(bool mute);
 
+    /* [optional] update the active backend resampler's output/input ratio.
+     * Called from the backend's playback callback and must be realtime safe. */
+    bool (*setRate)(double ratio);
+
     /* return the current total playback latency in microseconds */
     uint64_t (*latency)(void);
   }