mirror of
https://github.com/mozilla/gecko-dev.git
synced 2024-11-08 04:27:37 +00:00
393 lines
12 KiB
HTML
393 lines
12 KiB
HTML
<!DOCTYPE HTML>
|
|
<html>
|
|
<head>
|
|
<title>Test the decodeAudioData API and Resampling</title>
|
|
<script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
|
|
<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
|
|
</head>
|
|
<body>
|
|
<pre id="test">
|
|
<script src="webaudio.js" type="text/javascript"></script>
|
|
<script type="text/javascript">
|
|
|
|
// These routines have been copied verbatim from WebKit, and are used in order
|
|
// to convert a memory buffer into a wave buffer.
|
|
function writeString(s, a, offset) {
|
|
for (var i = 0; i < s.length; ++i) {
|
|
a[offset + i] = s.charCodeAt(i);
|
|
}
|
|
}
|
|
|
|
function writeInt16(n, a, offset) {
|
|
n = Math.floor(n);
|
|
|
|
var b1 = n & 255;
|
|
var b2 = (n >> 8) & 255;
|
|
|
|
a[offset + 0] = b1;
|
|
a[offset + 1] = b2;
|
|
}
|
|
|
|
function writeInt32(n, a, offset) {
|
|
n = Math.floor(n);
|
|
var b1 = n & 255;
|
|
var b2 = (n >> 8) & 255;
|
|
var b3 = (n >> 16) & 255;
|
|
var b4 = (n >> 24) & 255;
|
|
|
|
a[offset + 0] = b1;
|
|
a[offset + 1] = b2;
|
|
a[offset + 2] = b3;
|
|
a[offset + 3] = b4;
|
|
}
|
|
|
|
function writeAudioBuffer(audioBuffer, a, offset) {
|
|
var n = audioBuffer.length;
|
|
var channels = audioBuffer.numberOfChannels;
|
|
|
|
for (var i = 0; i < n; ++i) {
|
|
for (var k = 0; k < channels; ++k) {
|
|
var buffer = audioBuffer.getChannelData(k);
|
|
var sample = buffer[i] * 32768.0;
|
|
|
|
// Clip samples to the limitations of 16-bit.
|
|
// If we don't do this then we'll get nasty wrap-around distortion.
|
|
if (sample < -32768)
|
|
sample = -32768;
|
|
if (sample > 32767)
|
|
sample = 32767;
|
|
|
|
writeInt16(sample, a, offset);
|
|
offset += 2;
|
|
}
|
|
}
|
|
}
|
|
|
|
function createWaveFileData(audioBuffer) {
|
|
var frameLength = audioBuffer.length;
|
|
var numberOfChannels = audioBuffer.numberOfChannels;
|
|
var sampleRate = audioBuffer.sampleRate;
|
|
var bitsPerSample = 16;
|
|
var byteRate = sampleRate * numberOfChannels * bitsPerSample/8;
|
|
var blockAlign = numberOfChannels * bitsPerSample/8;
|
|
var wavDataByteLength = frameLength * numberOfChannels * 2; // 16-bit audio
|
|
var headerByteLength = 44;
|
|
var totalLength = headerByteLength + wavDataByteLength;
|
|
|
|
var waveFileData = new Uint8Array(totalLength);
|
|
|
|
var subChunk1Size = 16; // for linear PCM
|
|
var subChunk2Size = wavDataByteLength;
|
|
var chunkSize = 4 + (8 + subChunk1Size) + (8 + subChunk2Size);
|
|
|
|
writeString("RIFF", waveFileData, 0);
|
|
writeInt32(chunkSize, waveFileData, 4);
|
|
writeString("WAVE", waveFileData, 8);
|
|
writeString("fmt ", waveFileData, 12);
|
|
|
|
writeInt32(subChunk1Size, waveFileData, 16); // SubChunk1Size (4)
|
|
writeInt16(1, waveFileData, 20); // AudioFormat (2)
|
|
writeInt16(numberOfChannels, waveFileData, 22); // NumChannels (2)
|
|
writeInt32(sampleRate, waveFileData, 24); // SampleRate (4)
|
|
writeInt32(byteRate, waveFileData, 28); // ByteRate (4)
|
|
writeInt16(blockAlign, waveFileData, 32); // BlockAlign (2)
|
|
writeInt32(bitsPerSample, waveFileData, 34); // BitsPerSample (4)
|
|
|
|
writeString("data", waveFileData, 36);
|
|
writeInt32(subChunk2Size, waveFileData, 40); // SubChunk2Size (4)
|
|
|
|
// Write actual audio data starting at offset 44.
|
|
writeAudioBuffer(audioBuffer, waveFileData, 44);
|
|
|
|
return waveFileData;
|
|
}
|
|
|
|
</script>
|
|
<script class="testbody" type="text/javascript">
|
|
|
|
SimpleTest.waitForExplicitFinish();
|
|
|
|
// fuzzTolerance and fuzzToleranceMobile are used to determine fuzziness
|
|
// thresholds. They're needed to make sure that we can deal with neglibible
|
|
// differences in the binary buffer caused as a result of resampling the
|
|
// audio. fuzzToleranceMobile is typically larger on mobile platforms since
|
|
// we do fixed-point resampling as opposed to floating-point resampling on
|
|
// those platforms.
|
|
var files = [
|
|
// An ogg file, 44.1khz, mono
|
|
{
|
|
url: "ting-44.1k-1ch.ogg",
|
|
valid: true,
|
|
expectedUrl: "ting-44.1k-1ch.wav",
|
|
numberOfChannels: 1,
|
|
frames: 30592,
|
|
sampleRate: 44100,
|
|
duration: 0.693,
|
|
fuzzTolerance: 5,
|
|
fuzzToleranceMobile: 1284
|
|
},
|
|
// An ogg file, 44.1khz, stereo
|
|
{
|
|
url: "ting-44.1k-2ch.ogg",
|
|
valid: true,
|
|
expectedUrl: "ting-44.1k-2ch.wav",
|
|
numberOfChannels: 2,
|
|
frames: 30592,
|
|
sampleRate: 44100,
|
|
duration: 0.693,
|
|
fuzzTolerance: 6,
|
|
fuzzToleranceMobile: 2544
|
|
},
|
|
// An ogg file, 48khz, mono
|
|
{
|
|
url: "ting-48k-1ch.ogg",
|
|
valid: true,
|
|
expectedUrl: "ting-48k-1ch.wav",
|
|
numberOfChannels: 1,
|
|
frames: 33297,
|
|
sampleRate: 48000,
|
|
duration: 0.693,
|
|
fuzzTolerance: 5,
|
|
fuzzToleranceMobile: 1388
|
|
},
|
|
// An ogg file, 48khz, stereo
|
|
{
|
|
url: "ting-48k-2ch.ogg",
|
|
valid: true,
|
|
expectedUrl: "ting-48k-2ch.wav",
|
|
numberOfChannels: 2,
|
|
frames: 33297,
|
|
sampleRate: 48000,
|
|
duration: 0.693,
|
|
fuzzTolerance: 14,
|
|
fuzzToleranceMobile: 2752
|
|
},
|
|
// Make sure decoding a wave file results in the same buffer (for both the
|
|
// resampling and non-resampling cases)
|
|
{
|
|
url: "ting-44.1k-1ch.wav",
|
|
valid: true,
|
|
expectedUrl: "ting-44.1k-1ch.wav",
|
|
numberOfChannels: 1,
|
|
frames: 30592,
|
|
sampleRate: 44100,
|
|
duration: 0.693,
|
|
fuzzTolerance: 0,
|
|
fuzzToleranceMobile: 0
|
|
},
|
|
{
|
|
url: "ting-48k-1ch.wav",
|
|
valid: true,
|
|
expectedUrl: "ting-48k-1ch.wav",
|
|
numberOfChannels: 1,
|
|
frames: 33297,
|
|
sampleRate: 48000,
|
|
duration: 0.693,
|
|
fuzzTolerance: 0,
|
|
fuzzToleranceMobile: 0
|
|
},
|
|
// // A wave file
|
|
// //{ url: "24bit-44khz.wav", valid: true, expectedUrl: "24bit-44khz-expected.wav" },
|
|
// A non-audio file
|
|
{ url: "invalid.txt", valid: false, sampleRate: 44100 },
|
|
// A webm file with no audio
|
|
{ url: "noaudio.webm", valid: false, sampleRate: 48000 },
|
|
// A video ogg file with audio
|
|
{
|
|
url: "audio.ogv",
|
|
valid: true,
|
|
expectedUrl: "audio-expected.wav",
|
|
numberOfChannels: 2,
|
|
sampleRate: 44100,
|
|
frames: 47680,
|
|
duration: 1.0807,
|
|
fuzzTolerance: 106,
|
|
fuzzToleranceMobile: 3482
|
|
}
|
|
];
|
|
|
|
// Returns true if the memory buffers are less different that |fuzz| bytes
|
|
function fuzzyMemcmp(buf1, buf2, fuzz) {
|
|
var result = true;
|
|
var difference = 0;
|
|
is(buf1.length, buf2.length, "same length");
|
|
for (var i = 0; i < buf1.length; ++i) {
|
|
if (Math.abs(buf1[i] - buf2[i])) {
|
|
++difference;
|
|
}
|
|
}
|
|
if (difference > fuzz) {
|
|
ok(false, "Expected at most " + fuzz + " bytes difference, found " + difference + " bytes");
|
|
}
|
|
return difference <= fuzz;
|
|
}
|
|
|
|
function getFuzzTolerance(test) {
|
|
var kIsMobile =
|
|
navigator.userAgent.indexOf("Mobile") != -1 || // b2g
|
|
navigator.userAgent.indexOf("Android") != -1; // android
|
|
return kIsMobile ? test.fuzzToleranceMobile : test.fuzzTolerance;
|
|
}
|
|
|
|
function bufferIsSilent(buffer) {
|
|
for (var i = 0; i < buffer.length; ++i) {
|
|
if (buffer.getChannelData(0)[i] != 0) {
|
|
return false;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
function checkAudioBuffer(buffer, test) {
|
|
if (buffer.numberOfChannels != test.numberOfChannels) {
|
|
is(buffer.numberOfChannels, test.numberOfChannels, "Correct number of channels");
|
|
return;
|
|
}
|
|
ok(Math.abs(buffer.duration - test.duration) < 1e-3, "Correct duration");
|
|
if (Math.abs(buffer.duration - test.duration) >= 1e-3) {
|
|
ok(false, "got: " + buffer.duration + ", expected: " + test.duration);
|
|
}
|
|
is(buffer.sampleRate, test.sampleRate, "Correct sample rate");
|
|
is(buffer.length, test.frames, "Correct length");
|
|
|
|
var wave = createWaveFileData(buffer);
|
|
ok(fuzzyMemcmp(wave, test.expectedWaveData, getFuzzTolerance(test)), "Received expected decoded data");
|
|
}
|
|
|
|
function checkResampledBuffer(buffer, test, callback) {
|
|
if (buffer.numberOfChannels != test.numberOfChannels) {
|
|
is(buffer.numberOfChannels, test.numberOfChannels, "Correct number of channels");
|
|
return;
|
|
}
|
|
ok(Math.abs(buffer.duration - test.duration) < 1e-3, "Correct duration");
|
|
if (Math.abs(buffer.duration - test.duration) >= 1e-3) {
|
|
ok(false, "got: " + buffer.duration + ", expected: " + test.duration);
|
|
}
|
|
// Take into account the resampling when checking the size
|
|
var expectedLength = test.frames * buffer.sampleRate / test.sampleRate;
|
|
ok(Math.abs(buffer.length - expectedLength) < 1.0, "Correct length", "got " + buffer.length + ", expected about " + expectedLength);
|
|
|
|
// Playback the buffer in the original context, to resample back to the
|
|
// original rate and compare with the decoded buffer without resampling.
|
|
cx = test.nativeContext;
|
|
var expected = cx.createBufferSource();
|
|
expected.buffer = test.expectedBuffer;
|
|
expected.start();
|
|
var inverse = cx.createGain();
|
|
inverse.gain.value = -1;
|
|
expected.connect(inverse);
|
|
inverse.connect(cx.destination);
|
|
var resampled = cx.createBufferSource();
|
|
resampled.buffer = buffer;
|
|
resampled.start();
|
|
// This stop should do nothing, but it tests for bug 937475
|
|
resampled.stop(test.frames / cx.sampleRate);
|
|
resampled.connect(cx.destination);
|
|
cx.oncomplete = function(e) {
|
|
ok(!bufferIsSilent(e.renderedBuffer), "Expect buffer not silent");
|
|
// Resampling will lose the highest frequency components, so we should
|
|
// pass the difference through a low pass filter. However, either the
|
|
// input files don't have significant high frequency components or the
|
|
// tolerance in compareBuffers() is too high to detect them.
|
|
compareBuffers(e.renderedBuffer,
|
|
cx.createBuffer(test.numberOfChannels,
|
|
test.frames, test.sampleRate));
|
|
callback();
|
|
}
|
|
cx.startRendering();
|
|
}
|
|
|
|
function runResampling(test, response, callback) {
|
|
var sampleRate = test.sampleRate == 44100 ? 48000 : 44100;
|
|
var cx = new OfflineAudioContext(1, 1, sampleRate);
|
|
cx.decodeAudioData(response, function onSuccess(asyncResult) {
|
|
is(asyncResult.sampleRate, sampleRate, "Correct sample rate");
|
|
syncResult = cx.createBuffer(response, false);
|
|
compareBuffers(syncResult, asyncResult);
|
|
|
|
checkResampledBuffer(asyncResult, test, callback);
|
|
}, function onFailure() {
|
|
ok(false, "Expected successful decode with resample");
|
|
callback();
|
|
});
|
|
}
|
|
|
|
function runTest(test, response, callback) {
|
|
var expectCallback = false;
|
|
var cx = new OfflineAudioContext(test.numberOfChannels || 1,
|
|
test.frames || 1, test.sampleRate);
|
|
cx.decodeAudioData(response, function onSuccess(asyncResult) {
|
|
ok(expectCallback, "Success callback should fire asynchronously");
|
|
ok(test.valid, "Did expect success for test " + test.url);
|
|
|
|
syncResult = cx.createBuffer(response, false);
|
|
compareBuffers(syncResult, asyncResult);
|
|
checkAudioBuffer(asyncResult, test);
|
|
|
|
test.expectedBuffer = asyncResult;
|
|
test.nativeContext = cx;
|
|
runResampling(test, response, callback);
|
|
}, function onFailure() {
|
|
ok(expectCallback, "Failure callback should fire asynchronously");
|
|
ok(!test.valid, "Did expect failure for test " + test.url);
|
|
callback();
|
|
});
|
|
expectCallback = true;
|
|
}
|
|
|
|
function loadTest(test, callback) {
|
|
var xhr = new XMLHttpRequest();
|
|
xhr.open("GET", test.url, true);
|
|
xhr.responseType = "arraybuffer";
|
|
xhr.onload = function() {
|
|
var getExpected = new XMLHttpRequest();
|
|
getExpected.open("GET", test.expectedUrl, true);
|
|
getExpected.responseType = "arraybuffer";
|
|
getExpected.onload = function() {
|
|
test.expectedWaveData = new Uint8Array(getExpected.response);
|
|
runTest(test, xhr.response, callback);
|
|
};
|
|
getExpected.send();
|
|
};
|
|
xhr.send();
|
|
}
|
|
|
|
function loadNextTest() {
|
|
if (files.length) {
|
|
loadTest(files.shift(), loadNextTest);
|
|
} else {
|
|
SimpleTest.finish();
|
|
}
|
|
}
|
|
|
|
// Run some simple tests first
|
|
function callbackShouldNeverRun() {
|
|
ok(false, "callback should not fire");
|
|
}
|
|
(function() {
|
|
var cx = new AudioContext();
|
|
expectTypeError(function() {
|
|
cx.decodeAudioData(null, callbackShouldNeverRun, callbackShouldNeverRun);
|
|
});
|
|
expectTypeError(function() {
|
|
cx.decodeAudioData(undefined, callbackShouldNeverRun, callbackShouldNeverRun);
|
|
});
|
|
expectTypeError(function() {
|
|
cx.decodeAudioData(123, callbackShouldNeverRun, callbackShouldNeverRun);
|
|
});
|
|
expectTypeError(function() {
|
|
cx.decodeAudioData("buffer", callbackShouldNeverRun, callbackShouldNeverRun);
|
|
});
|
|
expectTypeError(function() {
|
|
cx.decodeAudioData(new Uint8Array(100), callbackShouldNeverRun, callbackShouldNeverRun);
|
|
});
|
|
})();
|
|
|
|
// Now, let's get real!
|
|
loadNextTest();
|
|
|
|
</script>
|
|
</pre>
|
|
</body>
|
|
</html>
|