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+<!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");
+
+    checkResampledBuffer(asyncResult, test, callback);
+  }, function onFailure() {
+    ok(false, "Expected successful decode with resample");
+    callback();
+  });
+}
+
+function runTest(test, response, callback) {
+  // We need to copy the array here, because decodeAudioData will detach the
+  // array's buffer.
+  var compressedAudio = response.slice(0);
+  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);
+
+    checkAudioBuffer(asyncResult, test);
+
+    test.expectedBuffer = asyncResult;
+    test.nativeContext = cx;
+    runResampling(test, compressedAudio, 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();
+  }
+}
+
+loadNextTest();
+
+</script>
+</pre>
+</body>
+</html>