我正在使用以下代码来解码来自nodejs套接字的音频块
window.AudioContext = window.AudioContext || window.webkitAudioContext;
var context = new AudioContext();
var delayTime = 0;
var init = 0;
var audioStack = [];
var nextTime = 0;
client.on('stream', function(stream, meta){
stream.on('data', function(data) {
context.decodeAudioData(data, function(buffer) {
audioStack.push(buffer);
if ((init!=0) || (audioStack.length > 10)) { // make sure we put at least 10 chunks in the buffer before starting
init++;
scheduleBuffers();
}
}, function(err) {
console.log("err(decodeAudioData): "+err);
});
});
});
function scheduleBuffers() {
while ( audioStack.length) {
var buffer = audioStack.shift();
var source = context.createBufferSource();
source.buffer = buffer;
source.connect(context.destination);
if (nextTime == 0)
nextTime = context.currentTime + 0.05; /// add 50ms latency to work well across systems - tune this if you like
source.start(nextTime);
nextTime+=source.buffer.duration; // Make the next buffer wait the length of the last buffer before being played
};
}
但是我无法弄清音频块之间的间隙/毛刺。
我也读过MediaSource,它可以做同样的事情,并且可以由播放器来处理时间,而不必手动进行。有人可以提供处理mp3数据的示例吗?
此外,使用Web音频API处理实时流媒体的正确方法是什么?我已经阅读了几乎所有与此主题相关的问题,并且似乎没有任何问题可以解决。有任何想法吗?
最佳答案
您可以以以下代码为例:https://github.com/kmoskwiak/node-tcp-streaming-server
它基本上使用媒体源扩展。您需要做的就是从视频转换为音频buffer = mediaSource.addSourceBuffer('audio/mpeg');