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刚开始一起推送AAC音频流,发现特别卡顿,调整了gstreamer为下面pipeline: appsrc name=appsrc ! decodebin ! audioconvert ! audioresample ! audio/x-raw,rate=48000 ! opusenc ! rtpopuspay timestamp-offset=0 pt=%d ! udpsink host=127.0.0.1 port=%d 音频是流畅了,可是视频还是卡顿,原音频是22050 Hz的。是不是目前AAC转OPus都不会很好效果?
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这个是一个demo用来验证, 可以看看srs srs已经很好的支持了rtmp 转 webrtc
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刚开始一起推送AAC音频流,发现特别卡顿,调整了gstreamer为下面pipeline:
appsrc name=appsrc ! decodebin ! audioconvert ! audioresample ! audio/x-raw,rate=48000 ! opusenc ! rtpopuspay timestamp-offset=0 pt=%d ! udpsink host=127.0.0.1 port=%d
音频是流畅了,可是视频还是卡顿,原音频是22050 Hz的。是不是目前AAC转OPus都不会很好效果?
The text was updated successfully, but these errors were encountered: