如何将 getUsermedia 音频流转换为 blob 或缓冲区? [英] how to convert getUsermedia audio stream into a blob or buffer?

问题描述

我正在从 getUserMeda 获取音频流，然后将其转换为 blob 或缓冲区，并在音频即将到来时将其发送到服务器我正在使用 socket.io 将其发送到服务器如何将音频媒体流转换为缓冲区?

I am getting audio stream from getUserMeda and then convert it into a blob or buffer and send it to server as audio is comming I am using socket.io to emit it to server how can i convert audio mediastream into buffer?

以下是我写的代码

navigator.getUserMedia({audio: true, video: false}, function(stream) {
webcamstream = stream;
var media = stream.getAudioTracks();
socket.emit("sendaudio", media);
},
function(e){
   console.log(e);
  }
});

如何将流转换为缓冲区并将其发送到 node.js 服务器，因为流来自 getusermedia 函数?

How to convert stream into buffer and emit it to node.js server as stream comes from getusermedia function?

推荐答案

根据@MuazKhan 的评论，使用 MediaRecorder(在 Firefox 中，最终将在 Chrome 中)或 RecordRTC/etc 将数据捕获到 blob 中.然后您可以通过以下几种方法之一将其导出到服务器进行分发:WebSockets、WebRTC DataChannels 等.请注意，这些不能保证实时传输数据，而且 MediaRecorder 还没有比特率控制.如果传输延迟，数据可能会在本地累积.

Per @MuazKhan's comment, use MediaRecorder (in Firefox, eventually will be in Chrome) or RecordRTC/etc to capture the data into blobs. Then you can export it via one of several methods to the server for distribution: WebSockets, WebRTC DataChannels, etc. Note that these are NOT guaranteed to transfer the data in realtime, and also MediaRecorder does not yet have bitrate controls. If transmission is delayed, data may build up locally.

如果实时(重新)传输很重要，强烈考虑使用 PeerConnection 代替服务器(根据@Robert 的评论)，然后将其转换为流.(如何完成取决于服务器，但您应该对 Opus 数据进行编码以重新打包或解码和重新编码.)虽然重新编码通常不好，但在这种情况下，您最好通过 NetEq 进行解码(webrtc.org 堆栈的抖动缓冲区和 PacketLossConcealment 代码)并获得干净的实时音频流，以重新编码流媒体，处理丢失和抖动.

If realtime (re)transmission is important, strongly consider using instead a PeerConnection to a server (per @Robert's comment) and then transform it there into a stream. (How that is done will depend on the server, but you should have encoded Opus data to either repackage or decode and re-encode.) While re-encoding is generally not good, in this case you would do best to decode through NetEq (webrtc.org stack's jitter-buffer and PacketLossConcealment code) and get a clean realtime audio stream to re-encode for streaming, with loss and jitter dealt with.

这篇关于如何将 getUsermedia 音频流转换为 blob 或缓冲区?的文章就介绍到这了，希望我们推荐的答案对大家有所帮助，也希望大家多多支持IT屋！