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