[英]AT&T Enhanced WebRTC + play .mp3
I'm trying to use AT&T's WebRTC JavaScript development framework. 我正在尝试使用AT&T的WebRTC JavaScript开发框架。 I don't see documentation for modifying a live call.
我看不到用于修改实时通话的文档。 Is it possible to modify a live call by playing an
.mp3
audio file? 是否可以通过播放
.mp3
音频文件来修改实时通话?
http://developer.att.com/enhanced-webrtc http://developer.att.com/enhanced-webrtc
I'm not familiar with AT&T framework, but you can simply add another AUDIO tag that you will play during the call whenever you want. 我对AT&T框架不熟悉,但是您可以简单地添加另一个AUDIO标签,该标签将在通话期间随时播放。 If you also want to interrupt the call, then you can simply mute the AUDIO/VIDEO tag with the remote stream attached.
如果您还想打断电话,则只需将带有远程流的AUDIO / VIDEO标签静音即可。
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