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Before this patch, if a send audio MediaStreamTrack ended, we ended up not sending anything over the network. If replaceTrack() at that point replaced the ended track with a live one, we'd start sending data again, but the rtp stream would continue from where the previous track ended. Having a gap in audio like that would confuse a receiver's *video* jitter buffer, because it's trying to sync to an audio track that just had a massive amount of "jitter" (it can't tell the difference). This patch fixes this by adding a track layer in MediaPipelineTransmit that remains active for as long as the MediaPipeline is active. Thus if the send audio MediaStreamTrack ends, we continue sending silence over the network, which the receiver can understand. If later replaced, the receiver sees real audio instead of silence and continues gracefully. Differential Revision: https://phabricator.services.mozilla.com/D56619 --HG-- extra : moz-landing-system : lando |
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| moz.build | ||
| webrtc_update.sh | ||
| webrtc_version.h | ||
| webrtc.mozbuild | ||