为什么在通过 WebRTC 从 Android 流式传输时视频分辨率会发生变化

Question

我正在尝试使用 WebRTC 在 Android 上从 Chrome 以 640x480 的分辨率进行流式传输，视频从那个开始，但随后分辨率下降到 320x240。

以下是发送的 getUserMedia 参数：

 "getUserMedia": [
  {
   "origin": "http://webrtc.example.com:3001",
   "pid": 30062,
   "rid": 15,
   "video": "mandatory: {minWidth:640, maxWidth:640, minHeight:480, maxHeight:480}"
  }

我的问题是为什么分辨率会下降？当我在我的 Mac 上从 Chrome 尝试时，并没有发生。我想进行调整，使视频分辨率不变。

video frames dumped using ffmpeg

chrome://webrtc-internals text dump

我正在使用 Licode WebRTC streaming server, but have also seen the same behavior using Kurento。

Answer 1

getUserMedia 约束只影响从浏览器请求到硬件并作为流返回的媒体。 getUserMedia 约束对之后对该流所做的操作没有任何影响（即，当它通过连接流式传输时）。您看到的降级发生在 PeerConnection 层，而不是 getUserMedia 层。当硬件和带宽统计表明性能低下时，由 webrtc 实现触发降级，并由双方协商。

[Hardware] <-   getUserMedia   -> [javascript client] <- PeerConnection -> [another client]
           <- 640x480 captured ->                     <-  320x240 sent  ->

您必须深入研究源代码以获取每个实现中如何完成的文档和证据，但对行为的引用：

来自OReilly Chapter on WebRTC：

The good news is that the WebRTC audio and video engines work together with the underlying network transport to probe the available bandwidth and optimize delivery of the media streams. However, DataChannel transfers require additional application logic: the application must monitor the amount of buffered data and be ready to adjust as needed.

...

WebRTC audio and video engines will dynamically adjust the bitrate of the media streams to match the conditions of the network link between the peers. The application can set and update the media constraints (e.g., video resolution, framerate, and so on), and the engines do the rest—this part is easy.

为什么在通过 WebRTC 从 Android 流式传输时视频分辨率会发生变化

Why does video resolution change when streaming from Android via WebRTC

video

android

google-chrome

video-processing

webrtc