Yes, of course this is possible! :)
- Make sure the audio fulfill CORS-requirements so we can load it with AJAX (loading from same origin as the page will of course fulfill this).
- Load the file as
ArrayBuffer
and decode it with AudioContext
- Calculate the number of segments and length of each (I use a time based length independent of channels below)
- Split the main buffer into smaller buffers
- Create a file-wrapper for the new buffer (below I made a simple WAVE wrapper for the demo)
- Feed that as Blob via an Object-URL to a new instance of the Audio element
- Keep keep track of the object-URLs so you can free them up when not needed anymore (
revokeObjectURL()
).
One drawback is of course that you would have to load the entire file into memory before processing it.
Example
Hopefully the file I'm using for the demo will be available through the current CDN that is used to allow CORS usage (I own the copyright, feel free to use it for testing, but only testing!! :) ). The loading and decoding can take some time depending on your system and connection, so please be patient...
Ideally you should use an asynchronous approach splitting the buffers, but the demo targets only the needed steps to make the buffer segments available as new file fragments.
Also note that I did not take into consideration the last segment to be shorter than the others (I use floor, you should use ceil for the segment count and cut the last block length short). I'll leave that as an exercise for the reader...
var actx = new(AudioContext || webkitAudioContext)(),
url = "//dl.dropboxusercontent.com/s/7ttdz6xsoaqbzdl/war_demo.mp3";
// STEP 1: Load audio file using AJAX ----------------------------------
fetch(url).then(function(resp) {return resp.arrayBuffer()}).then(decode);
// STEP 2: Decode the audio file ---------------------------------------
function decode(buffer) {
actx.decodeAudioData(buffer, split);
}
// STEP 3: Split the buffer --------------------------------------------
function split(abuffer) {
// calc number of segments and segment length
var channels = abuffer.numberOfChannels,
duration = abuffer.duration,
rate = abuffer.sampleRate,
segmentLen = 10,
count = Math.floor(duration / segmentLen),
offset = 0,
block = 10 * rate;
while(count--) {
var url = URL.createObjectURL(bufferToWave(abuffer, offset, block));
var audio = new Audio(url);
audio.controls = true;
audio.volume = 0.75;
document.body.appendChild(audio);
offset += block;
}
}
// Convert a audio-buffer segment to a Blob using WAVE representation
function bufferToWave(abuffer, offset, len) {
var numOfChan = abuffer.numberOfChannels,
length = len * numOfChan * 2 + 44,
buffer = new ArrayBuffer(length),
view = new DataView(buffer),
channels = [], i, sample,
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); // update data chunk
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;
}
}
audio {display:block;margin-bottom:1px}
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