问题描述
大多数Mediastream示例都是webCam-stream的解释。但是我需要从本地视频文件（.webm或mp4）创建MediaStream。请告诉我。
Most of the Mediastream examples are explanation by webCam-stream.but I need to create MediaStream from local videofile(.webm or mp4).Please tell me.
推荐答案
2017年5月4日更新： captureStream 现在Chrome和Firefox都支持API。
Updated at May 04, 2017: captureStream API are now supported both on Chrome and Firefox.
var stream_from_WebM_or_Mp4_File = videoTag.captureStream();
var stream_from_Canvas2D = canvasTag.captureStream(25);
参数25是帧率。
现在，您可以使用RTCPeerConnection API或使用MediaRecorder API记录来共享结果流。
Now you can share the resulting stream using either RTCPeerConnection API or record using MediaRecorder API.
请检查类似的答案：
有两种可能性：
1）captureStreamUntilEnded /
仅在Firefox上支持 mozCaptureStreamUntilEnded 。
It is supported only on Firefox as "mozCaptureStreamUntilEnded".
2）MediaSource API
chrome和firefox都支持MediaSource API;但是，它不是实时媒体流。
MediaSource APIs are supported both on chrome and firefox; however, it isn't a realtime media-stream.
你可以做的是读取文件块;使用WebSockets，socket.io或WebRTC数据通道等任何传输网关与其他用户共享;然后使用MediaSource API尽快播放这些块，而不是等待整个文件共享。
What you can do is read file chunks; share them with other users using any transmission gateway like WebSockets, socket.io or WebRTC datachannels; then use MediaSource API to play those chunks as soon as possible instead of waiting from entire file to be shared.
记住，关于铬和壁虎的WebRTC实现支持单一但实时的媒体源;这意味着您无法使用预先录制的媒体作为LIVE媒体源的捕获流。此外，您不能使用假的WebAudio流作为LIVE媒体源。
Remember, WebRTC implementations both on chromium and gecko supports single but "live" media source in the moment; it means that you can't use captured-stream from pre-recorded media as LIVE media source. Also, you can't use fake WebAudio stream as LIVE media source.
以下代码不适用于Firefox：
Following code will NOT work on Firefox:
preRecordedMediaStream = preRecordedMedia.mozCaptureStreamUntilEnded();
peer.addStream(preRecordedMediaStream);
您可以测试演示。
您可以使用FileReader / WebAudio API捕获预先录制的mp3 / ogg文件，并将其作为LIVE音频源共享给WebRTC对等连接，与我在 / 源代码。
You can capture pre-recorded mp3/ogg file using FileReader/WebAudio API and share as LIVE audio source to WebRTC peer connections, same as I did in this demo / source code.
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