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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
commit6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch)
treea68f146d7fa01f0134297619fbe7e33db084e0aa /testing/web-platform/tests/webaudio/resources/audit-util.js
parentInitial commit. (diff)
downloadthunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz
thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
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+// Copyright 2016 The Chromium Authors
+// 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;
+}