用户上传的MP3音频剪裁并只播放一部分实例页面

HTML代码：

<form>
    <input type="file" id="file" accept="audio/mpeg">
</form>

<p><audio id="audio" controls></audio></p>
                

JS代码：

file.onchange = function (event) {
    var target = event.target;
    var file = target.files[0];
    var type = file.type;
    // 开始识别
    var reader = new FileReader();
    reader.onload = function (event) {
        var arrBuffer = event.target.result;

        var audioCtx = new AudioContext();

        audioCtx.decodeAudioData(arrBuffer, function(audioBuffer) {
            var duration = audioBuffer.duration;
            var channels = audioBuffer.numberOfChannels;
            var rate = audioBuffer.sampleRate;

            // 3秒
            var startOffset = 0;
            var endOffset = rate * 3;
            var frameCount = endOffset - startOffset;
            var newAudioBuffer;

            newAudioBuffer = new AudioContext().createBuffer(channels, endOffset - startOffset, rate);
            var anotherArray = new Float32Array(frameCount);
            var offset = 0;

            for (var channel = 0; channel < channels; channel++) {
                audioBuffer.copyFromChannel(anotherArray, channel, startOffset);
                newAudioBuffer.copyToChannel(anotherArray, channel, offset);
            }

/**
* 直接播放使用下面的代码
// 创建AudioBufferSourceNode对象
var source = audioCtx.createBufferSource();
// 设置AudioBufferSourceNode对象的buffer为复制的3秒AudioBuffer对象
source.buffer = newAudioBuffer;
// 这一句是必须的，表示结束，没有这一句没法播放，没有声音
// 这里直接结束，实际上可以对结束做一些特效处理
source.connect(audioCtx.destination);
// 资源开始播放
source.start();
*/

            var blob = bufferToWave(newAudioBuffer, frameCount);
/**
* 转换成Base64使用下面的代码
var reader2 = new FileReader();
reader2.onload = function(evt){
    audio.src = evt.target.result;
};
reader2.readAsDataURL(blob);
*/
            // 使用Blob地址
            audio.src = URL.createObjectURL(blob);
        });
    };
    reader.readAsArrayBuffer(file);

};

// Convert AudioBuffer to a Blob using WAVE representation
function bufferToWave(abuffer, len) {
    var numOfChan = abuffer.numberOfChannels,
    length = len * numOfChan * 2 + 44,
    buffer = new ArrayBuffer(length),
    view = new DataView(buffer),
    channels = [], i, sample,
    offset = 0,
    pos = 0;

    // write WAVE header
    setUint32(0x46464952);                         // "RIFF"
    setUint32(length - 8);                         // file length - 8
    setUint32(0x45564157);                         // "WAVE"

    setUint32(0x20746d66);                         // "fmt " chunk
    setUint32(16);                                 // length = 16
    setUint16(1);                                  // PCM (uncompressed)
    setUint16(numOfChan);
    setUint32(abuffer.sampleRate);
    setUint32(abuffer.sampleRate * 2 * numOfChan); // avg. bytes/sec
    setUint16(numOfChan * 2);                      // block-align
    setUint16(16);                                 // 16-bit (hardcoded in this demo)

    setUint32(0x61746164);                         // "data" - chunk
    setUint32(length - pos - 4);                   // chunk length

    // write interleaved data
    for(i = 0; i < abuffer.numberOfChannels; i++)
        channels.push(abuffer.getChannelData(i));

    while(pos < length) {
        for(i = 0; i < numOfChan; i++) {             // interleave channels
            sample = Math.max(-1, Math.min(1, channels[i][offset])); // clamp
            sample = (0.5 + sample < 0 ? sample * 32768 : sample * 32767)|0; // scale to 16-bit signed int
            view.setInt16(pos, sample, true);          // write 16-bit sample
            pos += 2;
        }
        offset++                                     // next source sample
    }

    // create Blob
    return new Blob([buffer], {type: "audio/wav"});

    function setUint16(data) {
        view.setUint16(pos, data, true);
        pos += 2;
    }

    function setUint32(data) {
        view.setUint32(pos, data, true);
        pos += 4;
    }
}