Question

我正在尝试使用Binaryjs和一个简单的HTML客户端构建音频流服务器，这是我的节点服务器设置：

var BinaryServer = require('binaryjs').BinaryServer;
var fs = require('fs');
// Start Binary.js server
var server = BinaryServer({port: 9000}); //,chunkSize:150000
// Wait for new user connections
server.on('connection', function(client){
  var file = fs.createReadStream(__dirname + '/testsong.mp3');
  client.send(file);
  var chunks = [];

  file.on('data', function(data){
    console.log(client._socket._socket._writableState.finished);
    if (chunks.length < 40) {
      chunks.push(data);
    } else {
      if (client._socket._socket._writableState.finished === false) {
        console.log("Send");
        client.send(Buffer.concat(chunks));
      }
      chunks = [];
    }   
  });
});

它加载40个块然后将它们发送回连接的用户。这是我的index.html

<html>
  <script src="binaryjs.js"></script>
  <script src="jquery.js"></script>
<head>
</head>
<body>
  <script>
  var client = new BinaryClient('ws://localhost:9000');
  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);

        //Not sure if this line works.
        //  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
    };
  }
  </script>
</body>
</html>

代码部分有效，歌曲开始播放，但不断停止，模糊。我认为问题在于，有些块会提前播放，有些则会播放到很晚。已经过测试，在发送之前更改了延迟并更改了chunkSize，chunkLength以及许多其他代码。

我还读过Web Audio API有几个错误，或者我不知道如何实现它，context.decodeAudioData（）总是会为一首歌引发相同数量的null错误。

有没有办法调整它以使其正常播放或我该怎么办？解码似乎部分工作，所以可能有更好的延迟（已经尝试改变它）。

如何使用节点

0 个答案: