[tiredIsa]

Lets an application clock remote audio from its own output device (syrnike13-app issue #164).

• livekit::setExternalAudioPlayout(bool) — stops/resumes the synthetic device's 10 ms timer pulls.
• livekit::pullAudioPlayout() — one synchronous 10 ms NeedMorePlayData on the caller's thread; remote track sinks (FreshAudioStream) complete before it returns.
• C ABI: livekit_ffi_set_external_audio_playout, livekit_ffi_pull_audio_playout.

Default behaviour is unchanged.

Verification (local, Windows):

• cargo test -p libwebrtc --lib external_playout_delivers_remote_frames_before_return: 100 immediate reads on the caller thread, 0 queue drops, timer resumes after disabling.
• cargo test -p livekit --lib rtc_engine::lk_runtime::tests: one shared factory/ADM per process.
• ctest -R AudioSourceTest: 8 passed.
• App call bench with this SDK: playout latency 33→22 ms idle, 72→22 ms under CPU load and while viewing a screen share; 0 dropped/discontinuous frames.

🤖 Generated with Claude Code

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• docs/fresh-audio-stream.md
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• patches/client-sdk-rust/0001-native-v2-runtime.patch
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