[client] audio: rework synchronization for low latency

Drive playback timing from audio sample positions instead of packet
arrival cadence. Use SPICE multimedia timestamps to anchor streams,
detect discontinuities, and estimate the long-term source rate. Ignore
their coarse phase corrections when steering the playback buffer.

Measure the device clock independently and wait for it to stabilize
before enabling a bounded, slew-limited phase controller. Align startup
buffering from logical producer and consumer positions so backend
startup reserves do not become persistent latency.

Size the buffer from the backend period, measured delivery jitter, and
resampler delay. Update PipeWire and PulseAudio to report their actual
startup requirements and presentation latency. Default to a 10 ms
device period with 4 ms of additional buffering, and make detailed
synchronization diagnostics opt-in through audio:debug.

This reduces startup and steady-state latency while compensating for
physical and virtual device clock drift without reacting to transport
jitter or SPICE timestamp quantization.

client/src/audio.h

@@ -32,7 +32,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
 void audio_playbackStop(void);
 void audio_playbackVolume(int channels, const uint16_t volume[]);
 void audio_playbackMute(bool mute);
-void audio_playbackData(uint8_t * data, size_t size);
+void audio_playbackData(uint8_t * data, size_t size, uint32_t time);
 
 bool audio_supportsRecord(void);
 void audio_recordStart(int channels, int sampleRate, PSAudioFormat format);