Skip to content
New issue

Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.

By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.

Already on GitHub? Sign in to your account

WebRTC: Add audio jitter buffer in rtc2rtmp. Increase the jitter buffer for audio. #3454

Open
xiaozhihong opened this issue Mar 7, 2023 · 3 comments
Assignees
Labels
TransByAI Translated by AI/GPT. WebRTC WebRTC, RTC2RTMP or RTMP2RTC.

Comments

@xiaozhihong
Copy link
Collaborator

xiaozhihong commented Mar 7, 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

@winlinvip 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
@winlinvip winlinvip added the WebRTC WebRTC, RTC2RTMP or RTMP2RTC. label Mar 11, 2023
@winlinvip
Copy link
Member

winlinvip commented Mar 11, 2023

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

@winlinvip 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
@winlinvip winlinvip added the TransByAI Translated by AI/GPT. label Jul 29, 2023
@green-cats
Copy link

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

@winlinvip
Copy link
Member

Patch is welcome.

Sign up for free to join this conversation on GitHub. Already have an account? Sign in to comment
Labels
TransByAI Translated by AI/GPT. WebRTC WebRTC, RTC2RTMP or RTMP2RTC.
Projects
None yet
Development

No branches or pull requests

3 participants