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ossrs / srs

SRS is a simple, high-efficiency, real-time media server supporting RTMP, WebRTC, HLS, HTTP-FLV, HTTP-TS, SRT, MPEG-DASH, and GB28181.
https://ossrs.io
MIT License
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WebRTC: Add audio jitter buffer in rtc2rtmp. Increase the jitter buffer for audio. #3454

Open xiaozhihong opened 1 year ago

xiaozhihong commented 1 year ago

The audio RTP packet receives in WebRTC publisher transcodes from OPUS to AAC directly when enabled rtc_to_rtmp option. But it may be out of order or arrive after retransmitting, so we need an audio jitter buffer to make it in order with minimal latency.

TRANS_BY_GPT3

winlinvip commented 1 year ago

If out of order, the audio stream will be corrupt?

green-cats commented 4 weeks ago

@winlinvip can you add, please? this is a real problem...

winlinvip commented 4 weeks ago

Patch is welcome.