WebRTC: Support config the bitrate of transcoding AAC to Opus. v5.0.167, v6.0.60#3515
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…67, v6.0.60 (#3515) --------- Co-authored-by: john <hondaxiao@tencent.com>
@chundonglinlin您好,我删除并使用git clone -b develop https://gitee.com/ossrs/srs.git 重装了最新的dev版本,但发现不管是rtmp到WebRTC还是srt到WebRTC,音频码率都还是保持默认值不随配置文件而变化,我配置文件里面写的192000,但实际只有大概64000,能帮忙看看问题所在吗,谢谢!下面是我的配置文件: Hello, I have deleted and reinstalled the latest dev version using git clone -b develop https://gitee.com/ossrs/srs.git. However, I found that regardless of whether it is RTMP to WebRTC or SRT to WebRTC, the audio bitrate still remains the default value and does not change with the configuration file. I wrote 192000 in my configuration file, but in reality, it's only about 64000. Could you help me see what the problem is? Thank you! Below is my configuration file: |
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RTC support config audio bitrate by
opus_bitrateoraac_bitrate.