Update web-platform-tests to revision 44702f2bc8ea98bc32b5b244f2fe63c6ce66d49d

tests/wpt/web-platform-tests/webaudio/resources/audit-util.js

@@ -0,0 +1,195 @@
+// Copyright 2016 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+
+/**
+ * @fileOverview  This file includes legacy utility functions for the layout
+ *                test.
+ */
+
+// How many frames in a WebAudio render quantum.
+let RENDER_QUANTUM_FRAMES = 128;
+
+// Compare two arrays (commonly extracted from buffer.getChannelData()) with
+// constraints:
+//   options.thresholdSNR: Minimum allowed SNR between the actual and expected
+//     signal. The default value is 10000.
+//   options.thresholdDiffULP: Maximum allowed difference between the actual
+//     and expected signal in ULP(Unit in the last place). The default is 0.
+//   options.thresholdDiffCount: Maximum allowed number of sample differences
+//     which exceeds the threshold. The default is 0.
+//   options.bitDepth: The expected result is assumed to come from an audio
+//     file with this number of bits of precision. The default is 16.
+function compareBuffersWithConstraints(should, actual, expected, options) {
+  if (!options)
+    options = {};
+
+  // Only print out the message if the lengths are different; the
+  // expectation is that they are the same, so don't clutter up the
+  // output.
+  if (actual.length !== expected.length) {
+    should(
+        actual.length === expected.length,
+        'Length of actual and expected buffers should match')
+        .beTrue();
+  }
+
+  let maxError = -1;
+  let diffCount = 0;
+  let errorPosition = -1;
+  let thresholdSNR = (options.thresholdSNR || 10000);
+
+  let thresholdDiffULP = (options.thresholdDiffULP || 0);
+  let thresholdDiffCount = (options.thresholdDiffCount || 0);
+
+  // By default, the bit depth is 16.
+  let bitDepth = (options.bitDepth || 16);
+  let scaleFactor = Math.pow(2, bitDepth - 1);
+
+  let noisePower = 0, signalPower = 0;
+
+  for (let i = 0; i < actual.length; i++) {
+    let diff = actual[i] - expected[i];
+    noisePower += diff * diff;
+    signalPower += expected[i] * expected[i];
+
+    if (Math.abs(diff) > maxError) {
+      maxError = Math.abs(diff);
+      errorPosition = i;
+    }
+
+    // The reference file is a 16-bit WAV file, so we will almost never get
+    // an exact match between it and the actual floating-point result.
+    if (Math.abs(diff) > scaleFactor)
+      diffCount++;
+  }
+
+  let snr = 10 * Math.log10(signalPower / noisePower);
+  let maxErrorULP = maxError * scaleFactor;
+
+  should(snr, 'SNR').beGreaterThanOrEqualTo(thresholdSNR);
+
+  should(
+      maxErrorULP,
+      options.prefix + ': Maximum difference (in ulp units (' + bitDepth +
+          '-bits))')
+      .beLessThanOrEqualTo(thresholdDiffULP);
+
+  should(diffCount, options.prefix + ': Number of differences between results')
+      .beLessThanOrEqualTo(thresholdDiffCount);
+}
+
+// Create an impulse in a buffer of length sampleFrameLength
+function createImpulseBuffer(context, sampleFrameLength) {
+  let audioBuffer =
+      context.createBuffer(1, sampleFrameLength, context.sampleRate);
+  let n = audioBuffer.length;
+  let dataL = audioBuffer.getChannelData(0);
+
+  for (let k = 0; k < n; ++k) {
+    dataL[k] = 0;
+  }
+  dataL[0] = 1;
+
+  return audioBuffer;
+}
+
+// Create a buffer of the given length with a linear ramp having values 0 <= x <
+// 1.
+function createLinearRampBuffer(context, sampleFrameLength) {
+  let audioBuffer =
+      context.createBuffer(1, sampleFrameLength, context.sampleRate);
+  let n = audioBuffer.length;
+  let dataL = audioBuffer.getChannelData(0);
+
+  for (let i = 0; i < n; ++i)
+    dataL[i] = i / n;
+
+  return audioBuffer;
+}
+
+// Create an AudioBuffer of length |sampleFrameLength| having a constant value
+// |constantValue|. If |constantValue| is a number, the buffer has one channel
+// filled with that value. If |constantValue| is an array, the buffer is created
+// wit a number of channels equal to the length of the array, and channel k is
+// filled with the k'th element of the |constantValue| array.
+function createConstantBuffer(context, sampleFrameLength, constantValue) {
+  let channels;
+  let values;
+
+  if (typeof constantValue === 'number') {
+    channels = 1;
+    values = [constantValue];
+  } else {
+    channels = constantValue.length;
+    values = constantValue;
+  }
+
+  let audioBuffer =
+      context.createBuffer(channels, sampleFrameLength, context.sampleRate);
+  let n = audioBuffer.length;
+
+  for (let c = 0; c < channels; ++c) {
+    let data = audioBuffer.getChannelData(c);
+    for (let i = 0; i < n; ++i)
+      data[i] = values[c];
+  }
+
+  return audioBuffer;
+}
+
+// Create a stereo impulse in a buffer of length sampleFrameLength
+function createStereoImpulseBuffer(context, sampleFrameLength) {
+  let audioBuffer =
+      context.createBuffer(2, sampleFrameLength, context.sampleRate);
+  let n = audioBuffer.length;
+  let dataL = audioBuffer.getChannelData(0);
+  let dataR = audioBuffer.getChannelData(1);
+
+  for (let k = 0; k < n; ++k) {
+    dataL[k] = 0;
+    dataR[k] = 0;
+  }
+  dataL[0] = 1;
+  dataR[0] = 1;
+
+  return audioBuffer;
+}
+
+// Convert time (in seconds) to sample frames.
+function timeToSampleFrame(time, sampleRate) {
+  return Math.floor(0.5 + time * sampleRate);
+}
+
+// Compute the number of sample frames consumed by noteGrainOn with
+// the specified |grainOffset|, |duration|, and |sampleRate|.
+function grainLengthInSampleFrames(grainOffset, duration, sampleRate) {
+  let startFrame = timeToSampleFrame(grainOffset, sampleRate);
+  let endFrame = timeToSampleFrame(grainOffset + duration, sampleRate);
+
+  return endFrame - startFrame;
+}
+
+// True if the number is not an infinity or NaN
+function isValidNumber(x) {
+  return !isNaN(x) && (x != Infinity) && (x != -Infinity);
+}
+
+// Compute the (linear) signal-to-noise ratio between |actual| and
+// |expected|.  The result is NOT in dB!  If the |actual| and
+// |expected| have different lengths, the shorter length is used.
+function computeSNR(actual, expected) {
+  let signalPower = 0;
+  let noisePower = 0;
+
+  let length = Math.min(actual.length, expected.length);
+
+  for (let k = 0; k < length; ++k) {
+    let diff = actual[k] - expected[k];
+    signalPower += expected[k] * expected[k];
+    noisePower += diff * diff;
+  }
+
+  return signalPower / noisePower;
+}