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by bsder
2978 days ago
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Um ... from the paper ... "6.1 Limitations of Salsify No audio. Salsify does not encode or transmit audio." Claiming that you beat a bunch of codecs that have synchronized audio (even though they disable it) is kind of misleading ... |
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If you wanted to add audio to Salsify, you would want to control a receiver-side video and audio buffer to reduce audio gaps and keep a/v in sync during periods of happy network, but this is unlikely to affect the system's ability to recover more quickly from glitches or to avoid building up in-network queues that delay audio and video alike. If you watch the video (or see Figure 6(f), Figure 7, and Figure 8), I don't think there's much reason to think audio can justify what the Chrome/webrtc.org codebase is doing -- WebRTC's frame delays are distributed over a broad range (so it's not like they're synchronized to some fixed timebase either) and are very high, especially in the seconds after a network glitch.
More to the point for our academic work, it would have been trivial to add shitty audio that made no difference to the metrics. The hard-but-necessary part is in designing an evaluation metric to assess (1) the qualify of the reconstructed audio (including how many gaps/rebuffering delays were there when the de-jitter buffer went dry), (2) the delay of the reconstructed audio, keeping in mind this is not constant over time, (3) the quality of the audio/video synchronization, which also will not be constant over time. Then measuring that in a fair way across Skype/Facetime/Hangouts/WebRTC/Salsify, and then trying to decide which compromise on those three axes is desirable. Somebody should do all that work at some point, but it's a major piece of work to bite off and pretty far from anything we've done so far.