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rtc2rtmp: Sometimes there may be a decoding failure in the audio, resulting in no audio frames in the rtmp. #2690
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Which browser is used for streaming? Can it be reliably reproduced? Specifically, which version is it, 4.0 or higher?
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The specific version is 4.0.161. From the error message, we deduce that in certain situations, the audio frame data has a length of 0. If such a packet is sent to the decoder, the decoder will consider it as the end of decoding, and sending subsequent audio frames will result in an EOF error. Currently, we have implemented a simple patch to protect against this issue, and the aforementioned error with the audio frames has not occurred again. Attached are the details of the patch.
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After solving this problem, the next phenomenon is that the audio and video of the rtmp stream are both noticeably lagging. I suspect it may be a timestamp issue and further analysis is needed.
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@jamken 👍 Thanks |
* fix ossrs#2690, 如果发现音频包为空,则不送解码,以免提前结束了解码上下文 * fix ossrs#2690,出错信息中,显示相关参数名称 Co-authored-by: jiankai <wangjiankai@cmhi.chinamobile.com>
Description'
Please ensure that the markdown structure is maintained.
In the rtc-rtmp scenario, sometimes there is a decoding failure in the audio, resulting in no audio frames in the rtmp. From the logs, it can be seen that the decoder reported an "eof" error, which caused the decoding to fail. However, sometimes it works fine, and it is unclear what causes this situation to occur.
SRS version:
4.0
SRS configuration is as follows:
Replay
How to replay bug?
Steps to reproduce the bug:
Expected behavior:
The expected behavior is that both the WebRTC stream and the RTMP stream should have consistent audio and smooth playback without any stuttering.
RTMP playback:
The AAC audio frames are transmitted normally, and the video playback is smooth.
TRANS_BY_GPT3
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