Benjamin Carlsson ef6c058202 Disable clock synchronization in the voice audio sink (#43704) ...

Set the GStreamer audio sink's `slave-method` to `none` instead of
`resample`, and disable sink synchronization with `set_sync(false)`.

GitOrigin-RevId: a74f3e6d70e0d817c75bd963b31f16c69c2732ab

2026-09-08 04:55:29 +00:00

..

audio_sink.rs

Disable clock synchronization in the voice audio sink (#43704)

2026-09-08 04:55:29 +00:00

audio_track_tests.rs

Send processed microphone audio over RTP in voice-host (#43090)

2026-09-05 20:02:55 +00:00

audio_track.rs

Add bounded incoming Opus RTP handling to the voice host (#43100)

2026-09-05 20:58:32 +00:00

capture_worker_tests.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

capture_worker.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

device_buffers_tests.rs

Add bounded GStreamer playback components to the voice host (#43244)

2026-09-06 16:02:27 +00:00

device_buffers.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

devices_tests.rs

Add bounded GStreamer playback components to the voice host (#43244)

2026-09-06 16:02:27 +00:00

devices_unavailable.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

devices.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

incoming_tests.rs

Add bounded incoming Opus RTP handling to the voice host (#43100)

2026-09-05 20:58:32 +00:00

incoming.rs

Add bounded incoming Opus RTP handling to the voice host (#43100)

2026-09-05 20:58:32 +00:00

main_tests.rs

Send processed microphone audio over RTP in voice-host (#43090)

2026-09-05 20:02:55 +00:00

main.rs

Add live WebRTC voice conversations to the TUI (#43581)

2026-09-07 20:25:30 +00:00

playback_tests.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

playback.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

playout_tests.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

playout.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

processing_tests.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

processing.rs

Connect voice-host RTP audio to speaker playback (#43248)

2026-09-06 16:40:33 +00:00

runtime.rs

Initialize the packaged GStreamer runtime in the voice host (#42631)

2026-09-03 22:53:45 +00:00

transport_network_tests.rs

Add WebRTC negotiation to the voice host (#42676)

2026-09-04 05:07:02 +00:00

transport_runtime_tests.rs

Add WebRTC negotiation to the voice host (#42676)

2026-09-04 05:07:02 +00:00

transport_runtime.rs

Add WebRTC negotiation to the voice host (#42676)

2026-09-04 05:07:02 +00:00

transport_tests.rs

Add live WebRTC voice conversations to the TUI (#43581)

2026-09-07 20:25:30 +00:00

transport.rs

Add live WebRTC voice conversations to the TUI (#43581)

2026-09-07 20:25:30 +00:00