webrtc - WebRTC 是否支持视频的自适应比特率流媒体？

Question

我正在使用 WebRTC 来开发我的应用程序之一。
WebRTC 是否原生支持视频数据包的自适应比特率流，目前尚不清楚？ VP8/VP9 是否支持自适应比特率编码？ bitrate_controller WebRTC 是 ABR 的实现吗？

任何人都可以对此多加说明吗？我发现没有确凿的证据表明 WebRTC 本身支持视频的自适应流。

Answer 1

基于在此网站上找到的 WebRTC 文档:https://hpbn.co/webrtc/#audio-opus-and-video-vp8-bitrates我找到了这个:

When requesting audio and video from the browser, pay careful attention to the size and quality of the streams. While the hardware may be capable of capturing HD quality streams, the CPU and bandwidth must be able to keep up! Current WebRTC implementations use Opus and VP8 codecs:

• The Opus codec is used for audio and supports constant and variable bitrate encoding and requires 6–510 Kbit/s of bandwidth. The good
  news is that the codec can switch seamlessly and adapt to variable
  bandwidth.
• The VP8 codec used for video encoding also requires 100–2,000+ Kbit/s of bandwidth, and the bitrate depends on the quality of the streams: 720p at 30 FPS: 1.0~2.0 Mbps 360p at 30 FPS: 0.5~1.0 Mbps 180p at 30 FPS: 0.1~0.5 Mbps

As a result, a single-party HD call can require up to 2.5+ Mbps of network bandwidth. Add a few more peers, and the quality must drop to account for the extra bandwidth and CPU, GPU, and memory processing requirements.

因此，据我所知，两个编解码器都会使音频和视频流适应可用带宽。希望这可以帮助。