How to prevent audio streaming from buffering data?
zsk008 opened this issue · comments
- Use srs_aac_raw_publish to push the stream and use srs_play to play it. Close the publish, connect using play, and print out a frame of data first.
String onStatus
Object (1 items)
Property 'code' String NetStream.Data.Start
[2019-01-18 17:26:01.35] Audio packet id=0/0.0/0.0, type=Audio, dts=44, pts=44, ndiff=0, diff=0, size=4, AAC(22KHz,16bit,Stereo,SH)
Why does SRS need to cache this frame of data? Can it be not cached? Because if we use srs_audio_raw_publish to push G711U stream, the encoding information will change. The player will first receive a frame of AAC data, and the subsequent data will be G711U. This will cause decoding issues.
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Tested nginx-rtmp-module, there is no issue with the test.
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I replied in issue 301: In this situation, if we use the rtmp protocol, the first frame dumped is always cache_sh_audio, which will affect the decoding afterwards. Now, I have fixed it.
if (ds && cache_sh_audio && (ret = consumer->enqueue(cache_sh_audio, atc, jitter_algorithm)) != ERROR_SUCCESS) {
srs_error("dispatch audio sequence header failed. ret=%d", ret);
return ret;
Commenting it out can solve this problem, but will it affect HLS? Is there a better solution?
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This is the cache for the sequence header, which can be cleared when stopping the streaming. Take some time to check if it is well compatible with the browser.
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