如何在 Web Audio API 中设置麦克风缓冲区大小以避免 1/3 秒延迟（这在 Java 中要快得多）？

Question

此代码适用于 Windows 7 的 Firefox。它在扬声器上播放转换后的麦克风。它由 javascript 代码转换为音频幅度从 -1 到 1 的数字，每秒多次。它有 1/3 秒的延迟，但相同的硬件在 java 中运行得更快，我的https://sourceforge.net/projects/jsoundcard 软件通过时序统计调整缓冲区使用。当 OS 进程线程优先级提高时，它们都会变得更快，但 java 仍然快得多。没有理由 Firefox 或 Web Audio API 应该有更多的延迟，即使它已成为浏览器中无法正常运行的规范，但随着 HTML5（这是一个正在开发的开放标准）的变化。 Web Audio API 中是否有调整麦克风缓冲区大小的地方？

var volume = .7;
var maxMicrophoneAmplitude = 1;
var soundFunc = function(ins, outs){
    var microphoneAmplitude = ins[0];
    var estimatedFramesPerSecond = 44100;
    var decaySeconds = .1;
    var decay = 1/(estimatedFramesPerSecond*decaySeconds);
    maxMicrophoneAmplitude = Math.max(maxMicrophoneAmplitude, microphoneAmplitude)*(1-decay);
    microphoneAmplitude/maxMicrophoneAmplitude * volume
    outs[0] = microphoneAmplitude/maxMicrophoneAmplitude * volume;
};
soundFunc.inSize = 1;
soundFunc.outSize = 1;
var inputNodes = {};
window.onload = function(){
    var context = new AudioContext();
    if(!navigator.getUserMedia && navigator['mozGetUserMedia']){
        navigator.getUserMedia = navigator['mozGetUserMedia'];
    }
    if(!navigator.getUserMedia && navigator['webkitGetUserMedia']){
        navigator.getUserMedia = navigator['webkitGetUserMedia'];
    }
    if(!navigator.getUserMedia && navigator['msGetUserMedia']){
        navigator.getUserMedia = navigator['msGetUserMedia'];
    }
    var framesPerBuffer = 1024;
    var scriptNode = context.createScriptProcessor(framesPerBuffer, 1, 1);
    scriptNode.onaudioprocess = function(audioProcessingEvent){
        var inputBuffer = audioProcessingEvent.inputBuffer;
        var outputBuffer = audioProcessingEvent.outputBuffer;
        var chansIn = [];
        var chansOut = [];
        for(var c=0; c<inputBuffer.numberOfChannels; c++){
            chansIn[c] = inputBuffer.getChannelData(c);
        }
        for(var c=0; c<outputBuffer.numberOfChannels; c++){
            chansOut[c] = outputBuffer.getChannelData(c);
        }
        var ins = new Float32Array(inputBuffer.numberOfChannels);
        var outs = new Float32Array(outputBuffer.numberOfChannels);
        for(var f=0; f<framesPerBuffer; f++){
            for(var c=0; c<inputBuffer.numberOfChannels; c++){
                ins[c] = chansIn[c][f];
            }
            soundFunc(ins, outs);
            for(var c=0; c<outputBuffer.numberOfChannels; c++){
                chansOut[c][f] = outs[c];
            }
        }       
    }
    if(navigator.getUserMedia){
        navigator.getUserMedia(
            {'audio':true},
            function(stream){
                var input = context.createMediaStreamSource(stream);
                //http://stackoverflow.com/questions/22860468/html5-microphone-capture-stops-after-5-seconds-in-firefox says to save a reference to avoid sound ending after 5 seconds
                inputNodes.mic = input;
                input.connect(scriptNode);
                scriptNode.connect(context.destination);
                input.start();
            },
            function(e){ alert('Error capturing audio. e='+e); }
        );
    }else alert('getUserMedia not supported in this browser.'); 
};

Answer 1

我确实相信（如果我错了，@cwilso 可能会纠正我）API 使用的模型是从节点中提取样本，而不是节点将样本推送到其后的节点。这意味着您在脚本处理器中设置的缓冲区大小是延迟的来源，而不是麦克风流的输入缓冲区大小。尝试将缓冲区大小设置得更低。

var framesPerBuffer = 256;
var scriptNode = context.createScriptProcessor(framesPerBuffer, 1, 1);