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使用Javascript监控WebRTC视频（媒体）流质量

[英]Monitor WebRTC video (media) stream quality in Javascript

原文 2014-07-30 11:17:55 0 1 javascript/ video-streaming/ html5-video/ webrtc

I'm using WebRTC to stream video between peers, but changes in network conditions for some clients often produce quality changes in the received video stream. 我正在使用WebRTC在同级之间流式传输视频，但是某些客户端的网络条件变化通常会在接收到的视频流中产生质量变化。 People blame the service for these quality drops and clearly I (the service) cannot do anything about their network conditions. 人们将服务质量下降归咎于服务，显然我（该服务）无法对其网络状况做任何事情。 But showing an indication that the quality dropped due to network conditions on the client's end would most likely alleviate this problem. 但是，表明由于客户端网络质量而导致质量下降的迹象最有可能缓解此问题。

I've been searching Google & Stackoverflow for a while now and haven't seen any questions related to quality detection of incoming audio or video stream. 我已经搜索Google和Stackoverflow已有一段时间了，并且没有看到与传入音频或视频流的质量检测有关的任何问题。 Is there a way to monitor the quality (current bitrate or dropped frames, anything) during the live stream? 有没有一种方法可以在实时流中监视质量（当前的比特率或丢帧，什么）？

1 个解决方案

The getStats() API is what you are looking for if you want to programmatically access information in real time. 如果要实时以编程方式访问信息，则需要使用getStats（） API。 webrtc-internals is a separate webpage, that is providing you more informations than getstats because it has access to chrome internals, but eventually most of those info will be made available to getstats so people can have access to them from within their app. webrtc-internals是一个单独的网页，它比getstats提供更多的信息，因为它可以访问chrome内部，但最终这些信息中的大部分将提供给getstats，以便人们可以从其应用程序中访问它们。