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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
The text was updated successfully, but these errors were encountered:
winlinvip
changed the title
Add audio jitter buffer in rtc2rtmp. 增加音频的jitter buffer
WebRTC: Add audio jitter buffer in rtc2rtmp. 增加音频的jitter buffer
Mar 11, 2023
If out of order, the audio stream will be corrupt?
winlinvip
changed the title
WebRTC: Add audio jitter buffer in rtc2rtmp. 增加音频的jitter buffer
WebRTC: Add audio jitter buffer in rtc2rtmp. Increase the jitter buffer for audio.
Jul 29, 2023
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
The text was updated successfully, but these errors were encountered: