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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /dom/media/webaudio/test/blink
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
Diffstat (limited to 'dom/media/webaudio/test/blink')
-rw-r--r--dom/media/webaudio/test/blink/README9
-rw-r--r--dom/media/webaudio/test/blink/audio-testing.js192
-rw-r--r--dom/media/webaudio/test/blink/biquad-filters.js368
-rw-r--r--dom/media/webaudio/test/blink/biquad-testing.js153
-rw-r--r--dom/media/webaudio/test/blink/convolution-testing.js182
-rw-r--r--dom/media/webaudio/test/blink/mochitest.ini23
-rw-r--r--dom/media/webaudio/test/blink/panner-model-testing.js210
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodeAllPass.html32
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodeAutomation.html351
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodeBandPass.html34
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodeGetFrequencyResponse.html261
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodeHighPass.html33
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodeHighShelf.html33
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodeLowPass.html34
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodeLowShelf.html34
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodeNotch.html33
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodePeaking.html34
-rw-r--r--dom/media/webaudio/test/blink/test_biquadFilterNodeTail.html76
-rw-r--r--dom/media/webaudio/test/blink/test_iirFilterNode.html467
-rw-r--r--dom/media/webaudio/test/blink/test_iirFilterNodeGetFrequencyResponse.html97
20 files changed, 2656 insertions, 0 deletions
diff --git a/dom/media/webaudio/test/blink/README b/dom/media/webaudio/test/blink/README
new file mode 100644
index 000000000..1d819221f
--- /dev/null
+++ b/dom/media/webaudio/test/blink/README
@@ -0,0 +1,9 @@
+This directory contains tests originally borrowed from the Blink Web Audio test
+suite.
+
+The process of borrowing tests from Blink is as follows:
+
+* Import the pristine file from the Blink repo, noting the revision in the
+ commit message.
+* Modify the test files to turn the LayoutTest into a mochitest-plain and add
+* them to the test suite in a separate commit.
diff --git a/dom/media/webaudio/test/blink/audio-testing.js b/dom/media/webaudio/test/blink/audio-testing.js
new file mode 100644
index 000000000..c66d32c7f
--- /dev/null
+++ b/dom/media/webaudio/test/blink/audio-testing.js
@@ -0,0 +1,192 @@
+if (window.testRunner)
+ testRunner.overridePreference("WebKitWebAudioEnabled", "1");
+
+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;
+}
+
+function createAudioData(audioBuffer) {
+ return createWaveFileData(audioBuffer);
+}
+
+function finishAudioTest(event) {
+ var audioData = createAudioData(event.renderedBuffer);
+ testRunner.setAudioData(audioData);
+ testRunner.notifyDone();
+}
+
+// Create an impulse in a buffer of length sampleFrameLength
+function createImpulseBuffer(context, sampleFrameLength) {
+ var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+ var n = audioBuffer.length;
+ var dataL = audioBuffer.getChannelData(0);
+
+ for (var 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) {
+ var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+ var n = audioBuffer.length;
+ var dataL = audioBuffer.getChannelData(0);
+
+ for (var i = 0; i < n; ++i)
+ dataL[i] = i / n;
+
+ return audioBuffer;
+}
+
+// Create a buffer of the given length having a constant value.
+function createConstantBuffer(context, sampleFrameLength, constantValue) {
+ var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+ var n = audioBuffer.length;
+ var dataL = audioBuffer.getChannelData(0);
+
+ for (var i = 0; i < n; ++i)
+ dataL[i] = constantValue;
+
+ return audioBuffer;
+}
+
+// Create a stereo impulse in a buffer of length sampleFrameLength
+function createStereoImpulseBuffer(context, sampleFrameLength) {
+ var audioBuffer = context.createBuffer(2, sampleFrameLength, context.sampleRate);
+ var n = audioBuffer.length;
+ var dataL = audioBuffer.getChannelData(0);
+ var dataR = audioBuffer.getChannelData(1);
+
+ for (var 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 start with
+// the specified |grainOffset|, |duration|, and |sampleRate|.
+function grainLengthInSampleFrames(grainOffset, duration, sampleRate) {
+ var startFrame = timeToSampleFrame(grainOffset, sampleRate);
+ var 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);
+}
+
+function shouldThrowTypeError(func, text) {
+ var ok = false;
+ try {
+ func();
+ } catch (e) {
+ if (e instanceof TypeError) {
+ ok = true;
+ }
+ }
+ if (ok) {
+ testPassed(text + " threw TypeError.");
+ } else {
+ testFailed(text + " should throw TypeError.");
+ }
+}
diff --git a/dom/media/webaudio/test/blink/biquad-filters.js b/dom/media/webaudio/test/blink/biquad-filters.js
new file mode 100644
index 000000000..06fff98b1
--- /dev/null
+++ b/dom/media/webaudio/test/blink/biquad-filters.js
@@ -0,0 +1,368 @@
+// Taken from WebKit/LayoutTests/webaudio/resources/biquad-filters.js
+
+// A biquad filter has a z-transform of
+// H(z) = (b0 + b1 / z + b2 / z^2) / (1 + a1 / z + a2 / z^2)
+//
+// The formulas for the various filters were taken from
+// http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt.
+
+
+// Lowpass filter.
+function createLowpassFilter(freq, q, gain) {
+ var b0;
+ var b1;
+ var b2;
+ var a0;
+ var a1;
+ var a2;
+
+ if (freq == 1) {
+ // The formula below works, except for roundoff. When freq = 1,
+ // the filter is just a wire, so hardwire the coefficients.
+ b0 = 1;
+ b1 = 0;
+ b2 = 0;
+ a0 = 1;
+ a1 = 0;
+ a2 = 0;
+ } else {
+ var w0 = Math.PI * freq;
+ var alpha = 0.5 * Math.sin(w0) / Math.pow(10, q / 20);
+ var cos_w0 = Math.cos(w0);
+
+ b0 = 0.5 * (1 - cos_w0);
+ b1 = 1 - cos_w0;
+ b2 = b0;
+ a0 = 1 + alpha;
+ a1 = -2.0 * cos_w0;
+ a2 = 1 - alpha;
+ }
+
+ return normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+}
+
+function createHighpassFilter(freq, q, gain) {
+ var b0;
+ var b1;
+ var b2;
+ var a1;
+ var a2;
+
+ if (freq == 1) {
+ // The filter is 0
+ b0 = 0;
+ b1 = 0;
+ b2 = 0;
+ a0 = 1;
+ a1 = 0;
+ a2 = 0;
+ } else if (freq == 0) {
+ // The filter is 1. Computation of coefficients below is ok, but
+ // there's a pole at 1 and a zero at 1, so round-off could make
+ // the filter unstable.
+ b0 = 1;
+ b1 = 0;
+ b2 = 0;
+ a0 = 1;
+ a1 = 0;
+ a2 = 0;
+ } else {
+ var w0 = Math.PI * freq;
+ var alpha = 0.5 * Math.sin(w0) / Math.pow(10, q / 20);
+ var cos_w0 = Math.cos(w0);
+
+ b0 = 0.5 * (1 + cos_w0);
+ b1 = -1 - cos_w0;
+ b2 = b0;
+ a0 = 1 + alpha;
+ a1 = -2.0 * cos_w0;
+ a2 = 1 - alpha;
+ }
+
+ return normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+}
+
+function normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2) {
+ var scale = 1 / a0;
+
+ return {b0 : b0 * scale,
+ b1 : b1 * scale,
+ b2 : b2 * scale,
+ a1 : a1 * scale,
+ a2 : a2 * scale};
+}
+
+function createBandpassFilter(freq, q, gain) {
+ var b0;
+ var b1;
+ var b2;
+ var a0;
+ var a1;
+ var a2;
+ var coef;
+
+ if (freq > 0 && freq < 1) {
+ var w0 = Math.PI * freq;
+ if (q > 0) {
+ var alpha = Math.sin(w0) / (2 * q);
+ var k = Math.cos(w0);
+
+ b0 = alpha;
+ b1 = 0;
+ b2 = -alpha;
+ a0 = 1 + alpha;
+ a1 = -2 * k;
+ a2 = 1 - alpha;
+
+ coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+ } else {
+ // q = 0, and frequency is not 0 or 1. The above formula has a
+ // divide by zero problem. The limit of the z-transform as q
+ // approaches 0 is 1, so set the filter that way.
+ coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ }
+ } else {
+ // When freq = 0 or 1, the z-transform is identically 0,
+ // independent of q.
+ coef = {b0 : 0, b1 : 0, b2 : 0, a1 : 0, a2 : 0}
+ }
+
+ return coef;
+}
+
+function createLowShelfFilter(freq, q, gain) {
+ // q not used
+ var b0;
+ var b1;
+ var b2;
+ var a0;
+ var a1;
+ var a2;
+ var coef;
+
+ var S = 1;
+ var A = Math.pow(10, gain / 40);
+
+ if (freq == 1) {
+ // The filter is just a constant gain
+ coef = {b0 : A * A, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ } else if (freq == 0) {
+ // The filter is 1
+ coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ } else {
+ var w0 = Math.PI * freq;
+ var alpha = 1 / 2 * Math.sin(w0) * Math.sqrt((A + 1 / A) * (1 / S - 1) + 2);
+ var k = Math.cos(w0);
+ var k2 = 2 * Math.sqrt(A) * alpha;
+ var Ap1 = A + 1;
+ var Am1 = A - 1;
+
+ b0 = A * (Ap1 - Am1 * k + k2);
+ b1 = 2 * A * (Am1 - Ap1 * k);
+ b2 = A * (Ap1 - Am1 * k - k2);
+ a0 = Ap1 + Am1 * k + k2;
+ a1 = -2 * (Am1 + Ap1 * k);
+ a2 = Ap1 + Am1 * k - k2;
+ coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+ }
+
+ return coef;
+}
+
+function createHighShelfFilter(freq, q, gain) {
+ // q not used
+ var b0;
+ var b1;
+ var b2;
+ var a0;
+ var a1;
+ var a2;
+ var coef;
+
+ var A = Math.pow(10, gain / 40);
+
+ if (freq == 1) {
+ // When freq = 1, the z-transform is 1
+ coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ } else if (freq > 0) {
+ var w0 = Math.PI * freq;
+ var S = 1;
+ var alpha = 0.5 * Math.sin(w0) * Math.sqrt((A + 1 / A) * (1 / S - 1) + 2);
+ var k = Math.cos(w0);
+ var k2 = 2 * Math.sqrt(A) * alpha;
+ var Ap1 = A + 1;
+ var Am1 = A - 1;
+
+ b0 = A * (Ap1 + Am1 * k + k2);
+ b1 = -2 * A * (Am1 + Ap1 * k);
+ b2 = A * (Ap1 + Am1 * k - k2);
+ a0 = Ap1 - Am1 * k + k2;
+ a1 = 2 * (Am1 - Ap1*k);
+ a2 = Ap1 - Am1 * k-k2;
+
+ coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+ } else {
+ // When freq = 0, the filter is just a gain
+ coef = {b0 : A * A, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ }
+
+ return coef;
+}
+
+function createPeakingFilter(freq, q, gain) {
+ var b0;
+ var b1;
+ var b2;
+ var a0;
+ var a1;
+ var a2;
+ var coef;
+
+ var A = Math.pow(10, gain / 40);
+
+ if (freq > 0 && freq < 1) {
+ if (q > 0) {
+ var w0 = Math.PI * freq;
+ var alpha = Math.sin(w0) / (2 * q);
+ var k = Math.cos(w0);
+
+ b0 = 1 + alpha * A;
+ b1 = -2 * k;
+ b2 = 1 - alpha * A;
+ a0 = 1 + alpha / A;
+ a1 = -2 * k;
+ a2 = 1 - alpha / A;
+
+ coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+ } else {
+ // q = 0, we have a divide by zero problem in the formulas
+ // above. But if we look at the z-transform, we see that the
+ // limit as q approaches 0 is A^2.
+ coef = {b0 : A * A, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ }
+ } else {
+ // freq = 0 or 1, the z-transform is 1
+ coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ }
+
+ return coef;
+}
+
+function createNotchFilter(freq, q, gain) {
+ var b0;
+ var b1;
+ var b2;
+ var a0;
+ var a1;
+ var a2;
+ var coef;
+
+ if (freq > 0 && freq < 1) {
+ if (q > 0) {
+ var w0 = Math.PI * freq;
+ var alpha = Math.sin(w0) / (2 * q);
+ var k = Math.cos(w0);
+
+ b0 = 1;
+ b1 = -2 * k;
+ b2 = 1;
+ a0 = 1 + alpha;
+ a1 = -2 * k;
+ a2 = 1 - alpha;
+ coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+ } else {
+ // When q = 0, we get a divide by zero above. The limit of the
+ // z-transform as q approaches 0 is 0, so set the coefficients
+ // appropriately.
+ coef = {b0 : 0, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ }
+ } else {
+ // When freq = 0 or 1, the z-transform is 1
+ coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ }
+
+ return coef;
+}
+
+function createAllpassFilter(freq, q, gain) {
+ var b0;
+ var b1;
+ var b2;
+ var a0;
+ var a1;
+ var a2;
+ var coef;
+
+ if (freq > 0 && freq < 1) {
+ if (q > 0) {
+ var w0 = Math.PI * freq;
+ var alpha = Math.sin(w0) / (2 * q);
+ var k = Math.cos(w0);
+
+ b0 = 1 - alpha;
+ b1 = -2 * k;
+ b2 = 1 + alpha;
+ a0 = 1 + alpha;
+ a1 = -2 * k;
+ a2 = 1 - alpha;
+ coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+ } else {
+ // q = 0
+ coef = {b0 : -1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ }
+ } else {
+ coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+ }
+
+ return coef;
+}
+
+function filterData(filterCoef, signal, len) {
+ var y = new Array(len);
+ var b0 = filterCoef.b0;
+ var b1 = filterCoef.b1;
+ var b2 = filterCoef.b2;
+ var a1 = filterCoef.a1;
+ var a2 = filterCoef.a2;
+
+ // Prime the pump. (Assumes the signal has length >= 2!)
+ y[0] = b0 * signal[0];
+ y[1] = b0 * signal[1] + b1 * signal[0] - a1 * y[0];
+
+ // Filter all of the signal that we have.
+ for (var k = 2; k < Math.min(signal.length, len); ++k) {
+ y[k] = b0 * signal[k] + b1 * signal[k-1] + b2 * signal[k-2] - a1 * y[k-1] - a2 * y[k-2];
+ }
+
+ // If we need to filter more, but don't have any signal left,
+ // assume the signal is zero.
+ for (var k = signal.length; k < len; ++k) {
+ y[k] = - a1 * y[k-1] - a2 * y[k-2];
+ }
+
+ return y;
+}
+
+// Map the filter type name to a function that computes the filter coefficents for the given filter
+// type.
+var filterCreatorFunction = {"lowpass": createLowpassFilter,
+ "highpass": createHighpassFilter,
+ "bandpass": createBandpassFilter,
+ "lowshelf": createLowShelfFilter,
+ "highshelf": createHighShelfFilter,
+ "peaking": createPeakingFilter,
+ "notch": createNotchFilter,
+ "allpass": createAllpassFilter};
+
+var filterTypeName = {"lowpass": "Lowpass filter",
+ "highpass": "Highpass filter",
+ "bandpass": "Bandpass filter",
+ "lowshelf": "Lowshelf filter",
+ "highshelf": "Highshelf filter",
+ "peaking": "Peaking filter",
+ "notch": "Notch filter",
+ "allpass": "Allpass filter"};
+
+function createFilter(filterType, freq, q, gain) {
+ return filterCreatorFunction[filterType](freq, q, gain);
+}
diff --git a/dom/media/webaudio/test/blink/biquad-testing.js b/dom/media/webaudio/test/blink/biquad-testing.js
new file mode 100644
index 000000000..795adf601
--- /dev/null
+++ b/dom/media/webaudio/test/blink/biquad-testing.js
@@ -0,0 +1,153 @@
+// Globals, to make testing and debugging easier.
+var context;
+var filter;
+var signal;
+var renderedBuffer;
+var renderedData;
+
+var sampleRate = 44100.0;
+var pulseLengthFrames = .1 * sampleRate;
+
+// Maximum allowed error for the test to succeed. Experimentally determined.
+var maxAllowedError = 5.9e-8;
+
+// This must be large enough so that the filtered result is
+// essentially zero. See comments for createTestAndRun.
+var timeStep = .1;
+
+// Maximum number of filters we can process (mostly for setting the
+// render length correctly.)
+var maxFilters = 5;
+
+// How long to render. Must be long enough for all of the filters we
+// want to test.
+var renderLengthSeconds = timeStep * (maxFilters + 1) ;
+
+var renderLengthSamples = Math.round(renderLengthSeconds * sampleRate);
+
+// Number of filters that will be processed.
+var nFilters;
+
+function createImpulseBuffer(context, length) {
+ var impulse = context.createBuffer(1, length, context.sampleRate);
+ var data = impulse.getChannelData(0);
+ for (var k = 1; k < data.length; ++k) {
+ data[k] = 0;
+ }
+ data[0] = 1;
+
+ return impulse;
+}
+
+
+function createTestAndRun(context, filterType, filterParameters) {
+ // To test the filters, we apply a signal (an impulse) to each of
+ // the specified filters, with each signal starting at a different
+ // time. The output of the filters is summed together at the
+ // output. Thus for filter k, the signal input to the filter
+ // starts at time k * timeStep. For this to work well, timeStep
+ // must be large enough for the output of each filter to have
+ // decayed to zero with timeStep seconds. That way the filter
+ // outputs don't interfere with each other.
+
+ nFilters = Math.min(filterParameters.length, maxFilters);
+
+ signal = new Array(nFilters);
+ filter = new Array(nFilters);
+
+ impulse = createImpulseBuffer(context, pulseLengthFrames);
+
+ // Create all of the signal sources and filters that we need.
+ for (var k = 0; k < nFilters; ++k) {
+ signal[k] = context.createBufferSource();
+ signal[k].buffer = impulse;
+
+ filter[k] = context.createBiquadFilter();
+ filter[k].type = filterType;
+ filter[k].frequency.value = context.sampleRate / 2 * filterParameters[k].cutoff;
+ filter[k].detune.value = (filterParameters[k].detune === undefined) ? 0 : filterParameters[k].detune;
+ filter[k].Q.value = filterParameters[k].q;
+ filter[k].gain.value = filterParameters[k].gain;
+
+ signal[k].connect(filter[k]);
+ filter[k].connect(context.destination);
+
+ signal[k].start(timeStep * k);
+ }
+
+ context.oncomplete = checkFilterResponse(filterType, filterParameters);
+ context.startRendering();
+}
+
+function addSignal(dest, src, destOffset) {
+ // Add src to dest at the given dest offset.
+ for (var k = destOffset, j = 0; k < dest.length, j < src.length; ++k, ++j) {
+ dest[k] += src[j];
+ }
+}
+
+function generateReference(filterType, filterParameters) {
+ var result = new Array(renderLengthSamples);
+ var data = new Array(renderLengthSamples);
+ // Initialize the result array and data.
+ for (var k = 0; k < result.length; ++k) {
+ result[k] = 0;
+ data[k] = 0;
+ }
+ // Make data an impulse.
+ data[0] = 1;
+
+ for (var k = 0; k < nFilters; ++k) {
+ // Filter an impulse
+ var detune = (filterParameters[k].detune === undefined) ? 0 : filterParameters[k].detune;
+ var frequency = filterParameters[k].cutoff * Math.pow(2, detune / 1200); // Apply detune, converting from Cents.
+
+ var filterCoef = createFilter(filterType,
+ frequency,
+ filterParameters[k].q,
+ filterParameters[k].gain);
+ var y = filterData(filterCoef, data, renderLengthSamples);
+
+ // Accumulate this filtered data into the final output at the desired offset.
+ addSignal(result, y, timeToSampleFrame(timeStep * k, sampleRate));
+ }
+
+ return result;
+}
+
+function checkFilterResponse(filterType, filterParameters) {
+ return function(event) {
+ renderedBuffer = event.renderedBuffer;
+ renderedData = renderedBuffer.getChannelData(0);
+
+ reference = generateReference(filterType, filterParameters);
+
+ var len = Math.min(renderedData.length, reference.length);
+
+ var success = true;
+
+ // Maximum error between rendered data and expected data
+ var maxError = 0;
+
+ // Sample offset where the maximum error occurred.
+ var maxPosition = 0;
+
+ // Number of infinities or NaNs that occurred in the rendered data.
+ var invalidNumberCount = 0;
+
+ ok(nFilters == filterParameters.length, "Test wanted " + filterParameters.length + " filters but only " + maxFilters + " allowed.");
+
+ compareChannels(renderedData, reference, len, 0, 0, true);
+
+ // Check for bad numbers in the rendered output too.
+ // There shouldn't be any.
+ for (var k = 0; k < len; ++k) {
+ if (!isValidNumber(renderedData[k])) {
+ ++invalidNumberCount;
+ }
+ }
+
+ ok(invalidNumberCount == 0, "Rendered output has " + invalidNumberCount + " infinities or NaNs.");
+ SimpleTest.finish();
+ }
+}
diff --git a/dom/media/webaudio/test/blink/convolution-testing.js b/dom/media/webaudio/test/blink/convolution-testing.js
new file mode 100644
index 000000000..98ff0c775
--- /dev/null
+++ b/dom/media/webaudio/test/blink/convolution-testing.js
@@ -0,0 +1,182 @@
+var sampleRate = 44100.0;
+
+var renderLengthSeconds = 8;
+var pulseLengthSeconds = 1;
+var pulseLengthFrames = pulseLengthSeconds * sampleRate;
+
+function createSquarePulseBuffer(context, sampleFrameLength) {
+ var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+
+ var n = audioBuffer.length;
+ var data = audioBuffer.getChannelData(0);
+
+ for (var i = 0; i < n; ++i)
+ data[i] = 1;
+
+ return audioBuffer;
+}
+
+// The triangle buffer holds the expected result of the convolution.
+// It linearly ramps up from 0 to its maximum value (at the center)
+// then linearly ramps down to 0. The center value corresponds to the
+// point where the two square pulses overlap the most.
+function createTrianglePulseBuffer(context, sampleFrameLength) {
+ var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+
+ var n = audioBuffer.length;
+ var halfLength = n / 2;
+ var data = audioBuffer.getChannelData(0);
+
+ for (var i = 0; i < halfLength; ++i)
+ data[i] = i + 1;
+
+ for (var i = halfLength; i < n; ++i)
+ data[i] = n - i - 1;
+
+ return audioBuffer;
+}
+
+function log10(x) {
+ return Math.log(x)/Math.LN10;
+}
+
+function linearToDecibel(x) {
+ return 20*log10(x);
+}
+
+// Verify that the rendered result is very close to the reference
+// triangular pulse.
+function checkTriangularPulse(rendered, reference) {
+ var match = true;
+ var maxDelta = 0;
+ var valueAtMaxDelta = 0;
+ var maxDeltaIndex = 0;
+
+ for (var i = 0; i < reference.length; ++i) {
+ var diff = rendered[i] - reference[i];
+ var x = Math.abs(diff);
+ if (x > maxDelta) {
+ maxDelta = x;
+ valueAtMaxDelta = reference[i];
+ maxDeltaIndex = i;
+ }
+ }
+
+ // allowedDeviationFraction was determined experimentally. It
+ // is the threshold of the relative error at the maximum
+ // difference between the true triangular pulse and the
+ // rendered pulse.
+ var allowedDeviationDecibels = -129.4;
+ var maxDeviationDecibels = linearToDecibel(maxDelta / valueAtMaxDelta);
+
+ if (maxDeviationDecibels <= allowedDeviationDecibels) {
+ testPassed("Triangular portion of convolution is correct.");
+ } else {
+ testFailed("Triangular portion of convolution is not correct. Max deviation = " + maxDeviationDecibels + " dB at " + maxDeltaIndex);
+ match = false;
+ }
+
+ return match;
+}
+
+// Verify that the rendered data is close to zero for the first part
+// of the tail.
+function checkTail1(data, reference, breakpoint) {
+ var isZero = true;
+ var tail1Max = 0;
+
+ for (var i = reference.length; i < reference.length + breakpoint; ++i) {
+ var mag = Math.abs(data[i]);
+ if (mag > tail1Max) {
+ tail1Max = mag;
+ }
+ }
+
+ // Let's find the peak of the reference (even though we know a
+ // priori what it is).
+ var refMax = 0;
+ for (var i = 0; i < reference.length; ++i) {
+ refMax = Math.max(refMax, Math.abs(reference[i]));
+ }
+
+ // This threshold is experimentally determined by examining the
+ // value of tail1MaxDecibels.
+ var threshold1 = -129.7;
+
+ var tail1MaxDecibels = linearToDecibel(tail1Max/refMax);
+ if (tail1MaxDecibels <= threshold1) {
+ testPassed("First part of tail of convolution is sufficiently small.");
+ } else {
+ testFailed("First part of tail of convolution is not sufficiently small: " + tail1MaxDecibels + " dB");
+ isZero = false;
+ }
+
+ return isZero;
+}
+
+// Verify that the second part of the tail of the convolution is
+// exactly zero.
+function checkTail2(data, reference, breakpoint) {
+ var isZero = true;
+ var tail2Max = 0;
+ // For the second part of the tail, the maximum value should be
+ // exactly zero.
+ var threshold2 = 0;
+ for (var i = reference.length + breakpoint; i < data.length; ++i) {
+ if (Math.abs(data[i]) > 0) {
+ isZero = false;
+ break;
+ }
+ }
+
+ if (isZero) {
+ testPassed("Rendered signal after tail of convolution is silent.");
+ } else {
+ testFailed("Rendered signal after tail of convolution should be silent.");
+ }
+
+ return isZero;
+}
+
+function checkConvolvedResult(trianglePulse) {
+ return function(event) {
+ var renderedBuffer = event.renderedBuffer;
+
+ var referenceData = trianglePulse.getChannelData(0);
+ var renderedData = renderedBuffer.getChannelData(0);
+
+ var success = true;
+
+ // Verify the triangular pulse is actually triangular.
+
+ success = success && checkTriangularPulse(renderedData, referenceData);
+
+ // Make sure that portion after convolved portion is totally
+ // silent. But round-off prevents this from being completely
+ // true. At the end of the triangle, it should be close to
+ // zero. If we go farther out, it should be even closer and
+ // eventually zero.
+
+ // For the tail of the convolution (where the result would be
+ // theoretically zero), we partition the tail into two
+ // parts. The first is the at the beginning of the tail,
+ // where we tolerate a small but non-zero value. The second part is
+ // farther along the tail where the result should be zero.
+
+ // breakpoint is the point dividing the first two tail parts
+ // we're looking at. Experimentally determined.
+ var breakpoint = 12800;
+
+ success = success && checkTail1(renderedData, referenceData, breakpoint);
+
+ success = success && checkTail2(renderedData, referenceData, breakpoint);
+
+ if (success) {
+ testPassed("Test signal was correctly convolved.");
+ } else {
+ testFailed("Test signal was not correctly convolved.");
+ }
+
+ finishJSTest();
+ }
+}
diff --git a/dom/media/webaudio/test/blink/mochitest.ini b/dom/media/webaudio/test/blink/mochitest.ini
new file mode 100644
index 000000000..28bceb3a4
--- /dev/null
+++ b/dom/media/webaudio/test/blink/mochitest.ini
@@ -0,0 +1,23 @@
+[DEFAULT]
+tags=msg
+tags = webaudio
+subsuite = media
+support-files =
+ biquad-filters.js
+ biquad-testing.js
+ ../webaudio.js
+
+[test_biquadFilterNodeAllPass.html]
+[test_biquadFilterNodeAutomation.html]
+skip-if = true # Known problems with Biquad automation, e.g. Bug 1155709
+[test_biquadFilterNodeBandPass.html]
+[test_biquadFilterNodeGetFrequencyResponse.html]
+[test_biquadFilterNodeHighPass.html]
+[test_biquadFilterNodeHighShelf.html]
+[test_biquadFilterNodeLowPass.html]
+[test_biquadFilterNodeLowShelf.html]
+[test_biquadFilterNodeNotch.html]
+[test_biquadFilterNodePeaking.html]
+[test_biquadFilterNodeTail.html]
+[test_iirFilterNode.html]
+[test_iirFilterNodeGetFrequencyResponse.html]
diff --git a/dom/media/webaudio/test/blink/panner-model-testing.js b/dom/media/webaudio/test/blink/panner-model-testing.js
new file mode 100644
index 000000000..45460e276
--- /dev/null
+++ b/dom/media/webaudio/test/blink/panner-model-testing.js
@@ -0,0 +1,210 @@
+var sampleRate = 48000.0;
+
+var numberOfChannels = 1;
+
+// Time step when each panner node starts.
+var timeStep = 0.001;
+
+// Length of the impulse signal.
+var pulseLengthFrames = Math.round(timeStep * sampleRate);
+
+// How many panner nodes to create for the test
+var nodesToCreate = 100;
+
+// Be sure we render long enough for all of our nodes.
+var renderLengthSeconds = timeStep * (nodesToCreate + 1);
+
+// These are global mostly for debugging.
+var context;
+var impulse;
+var bufferSource;
+var panner;
+var position;
+var time;
+
+var renderedBuffer;
+var renderedLeft;
+var renderedRight;
+
+function createGraph(context, nodeCount) {
+ bufferSource = new Array(nodeCount);
+ panner = new Array(nodeCount);
+ position = new Array(nodeCount);
+ time = new Array(nodeCount);
+ // Angle between panner locations. (nodeCount - 1 because we want
+ // to include both 0 and 180 deg.
+ var angleStep = Math.PI / (nodeCount - 1);
+
+ if (numberOfChannels == 2) {
+ impulse = createStereoImpulseBuffer(context, pulseLengthFrames);
+ }
+ else
+ impulse = createImpulseBuffer(context, pulseLengthFrames);
+
+ for (var k = 0; k < nodeCount; ++k) {
+ bufferSource[k] = context.createBufferSource();
+ bufferSource[k].buffer = impulse;
+
+ panner[k] = context.createPanner();
+ panner[k].panningModel = "equalpower";
+ panner[k].distanceModel = "linear";
+
+ var angle = angleStep * k;
+ position[k] = {angle : angle, x : Math.cos(angle), z : Math.sin(angle)};
+ panner[k].positionX.value = position[k].x;
+ panner[k].positionZ.value = position[k].z;
+
+ bufferSource[k].connect(panner[k]);
+ panner[k].connect(context.destination);
+
+ // Start the source
+ time[k] = k * timeStep;
+ bufferSource[k].start(time[k]);
+ }
+}
+
+function createTestAndRun(context, nodeCount, numberOfSourceChannels) {
+ numberOfChannels = numberOfSourceChannels;
+
+ createGraph(context, nodeCount);
+
+ context.oncomplete = checkResult;
+ context.startRendering();
+}
+
+// Map our position angle to the azimuth angle (in degrees).
+//
+// An angle of 0 corresponds to an azimuth of 90 deg; pi, to -90 deg.
+function angleToAzimuth(angle) {
+ return 90 - angle * 180 / Math.PI;
+}
+
+// The gain caused by the EQUALPOWER panning model
+function equalPowerGain(angle) {
+ var azimuth = angleToAzimuth(angle);
+
+ if (numberOfChannels == 1) {
+ var panPosition = (azimuth + 90) / 180;
+
+ var gainL = Math.cos(0.5 * Math.PI * panPosition);
+ var gainR = Math.sin(0.5 * Math.PI * panPosition);
+
+ return { left : gainL, right : gainR };
+ } else {
+ if (azimuth <= 0) {
+ var panPosition = (azimuth + 90) / 90;
+
+ var gainL = 1 + Math.cos(0.5 * Math.PI * panPosition);
+ var gainR = Math.sin(0.5 * Math.PI * panPosition);
+
+ return { left : gainL, right : gainR };
+ } else {
+ var panPosition = azimuth / 90;
+
+ var gainL = Math.cos(0.5 * Math.PI * panPosition);
+ var gainR = 1 + Math.sin(0.5 * Math.PI * panPosition);
+
+ return { left : gainL, right : gainR };
+ }
+ }
+}
+
+function checkResult(event) {
+ renderedBuffer = event.renderedBuffer;
+ renderedLeft = renderedBuffer.getChannelData(0);
+ renderedRight = renderedBuffer.getChannelData(1);
+
+ // The max error we allow between the rendered impulse and the
+ // expected value. This value is experimentally determined. Set
+ // to 0 to make the test fail to see what the actual error is.
+ var maxAllowedError = 1.3e-6;
+
+ var success = true;
+
+ // Number of impulses found in the rendered result.
+ var impulseCount = 0;
+
+ // Max (relative) error and the index of the maxima for the left
+ // and right channels.
+ var maxErrorL = 0;
+ var maxErrorIndexL = 0;
+ var maxErrorR = 0;
+ var maxErrorIndexR = 0;
+
+ // Number of impulses that don't match our expected locations.
+ var timeCount = 0;
+
+ // Locations of where the impulses aren't at the expected locations.
+ var timeErrors = new Array();
+
+ for (var k = 0; k < renderedLeft.length; ++k) {
+ // We assume that the left and right channels start at the same instant.
+ if (renderedLeft[k] != 0 || renderedRight[k] != 0) {
+ // The expected gain for the left and right channels.
+ var pannerGain = equalPowerGain(position[impulseCount].angle);
+ var expectedL = pannerGain.left;
+ var expectedR = pannerGain.right;
+
+ // Absolute error in the gain.
+ var errorL = Math.abs(renderedLeft[k] - expectedL);
+ var errorR = Math.abs(renderedRight[k] - expectedR);
+
+ if (Math.abs(errorL) > maxErrorL) {
+ maxErrorL = Math.abs(errorL);
+ maxErrorIndexL = impulseCount;
+ }
+ if (Math.abs(errorR) > maxErrorR) {
+ maxErrorR = Math.abs(errorR);
+ maxErrorIndexR = impulseCount;
+ }
+
+ // Keep track of the impulses that didn't show up where we
+ // expected them to be.
+ var expectedOffset = timeToSampleFrame(time[impulseCount], sampleRate);
+ if (k != expectedOffset) {
+ timeErrors[timeCount] = { actual : k, expected : expectedOffset};
+ ++timeCount;
+ }
+ ++impulseCount;
+ }
+ }
+
+ if (impulseCount == nodesToCreate) {
+ testPassed("Number of impulses matches the number of panner nodes.");
+ } else {
+ testFailed("Number of impulses is incorrect. (Found " + impulseCount + " but expected " + nodesToCreate + ")");
+ success = false;
+ }
+
+ if (timeErrors.length > 0) {
+ success = false;
+ testFailed(timeErrors.length + " timing errors found in " + nodesToCreate + " panner nodes.");
+ for (var k = 0; k < timeErrors.length; ++k) {
+ testFailed("Impulse at sample " + timeErrors[k].actual + " but expected " + timeErrors[k].expected);
+ }
+ } else {
+ testPassed("All impulses at expected offsets.");
+ }
+
+ if (maxErrorL <= maxAllowedError) {
+ testPassed("Left channel gain values are correct.");
+ } else {
+ testFailed("Left channel gain values are incorrect. Max error = " + maxErrorL + " at time " + time[maxErrorIndexL] + " (threshold = " + maxAllowedError + ")");
+ success = false;
+ }
+
+ if (maxErrorR <= maxAllowedError) {
+ testPassed("Right channel gain values are correct.");
+ } else {
+ testFailed("Right channel gain values are incorrect. Max error = " + maxErrorR + " at time " + time[maxErrorIndexR] + " (threshold = " + maxAllowedError + ")");
+ success = false;
+ }
+
+ if (success) {
+ testPassed("EqualPower panner test passed");
+ } else {
+ testFailed("EqualPower panner test failed");
+ }
+
+ finishJSTest();
+}
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeAllPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeAllPass.html
new file mode 100644
index 000000000..266521c52
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeAllPass.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode All Pass Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ // Create offline audio context.
+ var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+ var filterParameters = [{cutoff : 0, q : 10, gain : 1 },
+ {cutoff : 1, q : 10, gain : 1 },
+ {cutoff : .5, q : 0, gain : 1 },
+ {cutoff : 0.25, q : 10, gain : 1 },
+ ];
+ createTestAndRun(context, "allpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeAutomation.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeAutomation.html
new file mode 100644
index 000000000..08ce71cce
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeAutomation.html
@@ -0,0 +1,351 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode All Pass Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ // Don't need to run these tests at high sampling rate, so just use a low one to reduce memory
+ // usage and complexity.
+ var sampleRate = 16000;
+
+ // How long to render for each test.
+ var renderDuration = 1;
+
+ // The definition of the linear ramp automation function.
+ function linearRamp(t, v0, v1, t0, t1) {
+ return v0 + (v1 - v0) * (t - t0) / (t1 - t0);
+ }
+
+ // Generate the filter coefficients for the specified filter using the given parameters for
+ // the given duration. |filterTypeFunction| is a function that returns the filter
+ // coefficients for one set of parameters. |parameters| is a property bag that contains the
+ // start and end values (as an array) for each of the biquad attributes. The properties are
+ // |freq|, |Q|, |gain|, and |detune|. |duration| is the number of seconds for which the
+ // coefficients are generated.
+ //
+ // A property bag with properties |b0|, |b1|, |b2|, |a1|, |a2|. Each propery is an array
+ // consisting of the coefficients for the time-varying biquad filter.
+ function generateFilterCoefficients(filterTypeFunction, parameters, duration) {
+ var endFrame = Math.ceil(duration * sampleRate);
+ var nCoef = endFrame;
+ var b0 = new Float64Array(nCoef);
+ var b1 = new Float64Array(nCoef);
+ var b2 = new Float64Array(nCoef);
+ var a1 = new Float64Array(nCoef);
+ var a2 = new Float64Array(nCoef);
+
+ var k = 0;
+ // If the property is not given, use the defaults.
+ var freqs = parameters.freq || [350, 350];
+ var qs = parameters.Q || [1, 1];
+ var gains = parameters.gain || [0, 0];
+ var detunes = parameters.detune || [0, 0];
+
+ for (var frame = 0; frame < endFrame; ++frame) {
+ // Apply linear ramp at frame |frame|.
+ var f = linearRamp(frame / sampleRate, freqs[0], freqs[1], 0, duration);
+ var q = linearRamp(frame / sampleRate, qs[0], qs[1], 0, duration);
+ var g = linearRamp(frame / sampleRate, gains[0], gains[1], 0, duration);
+ var d = linearRamp(frame / sampleRate, detunes[0], detunes[1], 0, duration);
+
+ // Compute actual frequency parameter
+ f = f * Math.pow(2, d / 1200);
+
+ // Compute filter coefficients
+ var coef = filterTypeFunction(f / (sampleRate / 2), q, g);
+ b0[k] = coef.b0;
+ b1[k] = coef.b1;
+ b2[k] = coef.b2;
+ a1[k] = coef.a1;
+ a2[k] = coef.a2;
+ ++k;
+ }
+
+ return {b0: b0, b1: b1, b2: b2, a1: a1, a2: a2};
+ }
+
+ // Apply the given time-varying biquad filter to the given signal, |signal|. |coef| should be
+ // the time-varying coefficients of the filter, as returned by |generateFilterCoefficients|.
+ function timeVaryingFilter(signal, coef) {
+ var length = signal.length;
+ // Use double precision for the internal computations.
+ var y = new Float64Array(length);
+
+ // Prime the pump. (Assumes the signal has length >= 2!)
+ y[0] = coef.b0[0] * signal[0];
+ y[1] = coef.b0[1] * signal[1] + coef.b1[1] * signal[0] - coef.a1[1] * y[0];
+
+ for (var n = 2; n < length; ++n) {
+ y[n] = coef.b0[n] * signal[n] + coef.b1[n] * signal[n-1] + coef.b2[n] * signal[n-2];
+ y[n] -= coef.a1[n] * y[n-1] + coef.a2[n] * y[n-2];
+ }
+
+ // But convert the result to single precision for comparison.
+ return y.map(Math.fround);
+ }
+
+ // Configure the audio graph using |context|. Returns the biquad filter node and the
+ // AudioBuffer used for the source.
+ function configureGraph(context, toneFrequency) {
+ // The source is just a simple sine wave.
+ var src = context.createBufferSource();
+ var b = context.createBuffer(1, renderDuration * sampleRate, sampleRate);
+ var data = b.getChannelData(0);
+ var omega = 2 * Math.PI * toneFrequency / sampleRate;
+ for (var k = 0; k < data.length; ++k) {
+ data[k] = Math.sin(omega * k);
+ }
+ src.buffer = b;
+ var f = context.createBiquadFilter();
+ src.connect(f);
+ f.connect(context.destination);
+
+ src.start();
+
+ return {filter: f, source: b};
+ }
+
+ function createFilterVerifier(filterCreator, threshold, parameters, input, message) {
+ return function (resultBuffer) {
+ var actual = resultBuffer.getChannelData(0);
+ var coefs = generateFilterCoefficients(filterCreator, parameters, renderDuration);
+
+ reference = timeVaryingFilter(input, coefs);
+
+ compareChannels(actual, reference);
+ };
+ }
+
+ var testPromises = [];
+
+ // Automate just the frequency parameter. A bandpass filter is used where the center
+ // frequency is swept across the source (which is a simple tone).
+ testPromises.push(function () {
+ var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+ // Center frequency of bandpass filter and also the frequency of the test tone.
+ var centerFreq = 10*440;
+
+ // Sweep the frequency +/- 9*440 Hz from the center. This should cause the output to low at
+ // the beginning and end of the test where the done is outside the pass band of the filter,
+ // but high in the center where the tone is near the center of the pass band.
+ var parameters = {
+ freq: [centerFreq - 9*440, centerFreq + 9*440]
+ }
+ var graph = configureGraph(context, centerFreq);
+ var f = graph.filter;
+ var b = graph.source;
+
+ f.type = "bandpass";
+ f.frequency.setValueAtTime(parameters.freq[0], 0);
+ f.frequency.linearRampToValueAtTime(parameters.freq[1], renderDuration);
+
+ return context.startRendering()
+ .then(createFilterVerifier(createBandpassFilter, 5e-5, parameters, b.getChannelData(0),
+ "Output of bandpass filter with frequency automation"));
+ }());
+
+ // Automate just the Q parameter. A bandpass filter is used where the Q of the filter is
+ // swept.
+ testPromises.push(function() {
+ var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+ // The frequency of the test tone.
+ var centerFreq = 440;
+
+ // Sweep the Q paramter between 1 and 200. This will cause the output of the filter to pass
+ // most of the tone at the beginning to passing less of the tone at the end. This is
+ // because we set center frequency of the bandpass filter to be slightly off from the actual
+ // tone.
+ var parameters = {
+ Q: [1, 200],
+ // Center frequency of the bandpass filter is just 25 Hz above the tone frequency.
+ freq: [centerFreq + 25, centerFreq + 25]
+ };
+ var graph = configureGraph(context, centerFreq);
+ var f = graph.filter;
+ var b = graph.source;
+
+ f.type = "bandpass";
+ f.frequency.value = parameters.freq[0];
+ f.Q.setValueAtTime(parameters.Q[0], 0);
+ f.Q.linearRampToValueAtTime(parameters.Q[1], renderDuration);
+
+ return context.startRendering()
+ .then(createFilterVerifier(createBandpassFilter, 1.4e-6, parameters, b.getChannelData(0),
+ "Output of bandpass filter with Q automation"));
+ }());
+
+ // Automate just the gain of the lowshelf filter. A test tone will be in the lowshelf part of
+ // the filter. The output will vary as the gain of the lowshelf is changed.
+ testPromises.push(function() {
+ var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+ // Frequency of the test tone.
+ var centerFreq = 440;
+
+ // Set the cutoff frequency of the lowshelf to be significantly higher than the test tone.
+ // Sweep the gain from 20 dB to -20 dB. (We go from 20 to -20 to easily verify that the
+ // filter didn't go unstable.)
+ var parameters = {
+ freq: [3500, 3500],
+ gain: [20, -20]
+ }
+ var graph = configureGraph(context, centerFreq);
+ var f = graph.filter;
+ var b = graph.source;
+
+ f.type = "lowshelf";
+ f.frequency.value = parameters.freq[0];
+ f.gain.setValueAtTime(parameters.gain[0], 0);
+ f.gain.linearRampToValueAtTime(parameters.gain[1], renderDuration);
+
+ context.startRendering()
+ .then(createFilterVerifier(createLowShelfFilter, 8e-6, parameters, b.getChannelData(0),
+ "Output of lowshelf filter with gain automation"));
+ }());
+
+ // Automate just the detune parameter. Basically the same test as for the frequncy parameter
+ // but we just use the detune parameter to modulate the frequency parameter.
+ testPromises.push(function() {
+ var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+ var centerFreq = 10*440;
+ var parameters = {
+ freq: [centerFreq, centerFreq],
+ detune: [-10*1200, 10*1200]
+ };
+ var graph = configureGraph(context, centerFreq);
+ var f = graph.filter;
+ var b = graph.source;
+
+ f.type = "bandpass";
+ f.frequency.value = parameters.freq[0];
+ f.detune.setValueAtTime(parameters.detune[0], 0);
+ f.detune.linearRampToValueAtTime(parameters.detune[1], renderDuration);
+
+ context.startRendering()
+ .then(createFilterVerifier(createBandpassFilter, 5e-6, parameters, b.getChannelData(0),
+ "Output of bandpass filter with detune automation"));
+ }());
+
+ // Automate all of the filter parameters at once. This is a basic check that everything is
+ // working. A peaking filter is used because it uses all of the parameters.
+ testPromises.push(function() {
+ var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+ var graph = configureGraph(context, 10*440);
+ var f = graph.filter;
+ var b = graph.source;
+
+ // Sweep all of the filter parameters. These are pretty much arbitrary.
+ var parameters = {
+ freq: [10000, 100],
+ Q: [f.Q.value, .0001],
+ gain: [f.gain.value, 20],
+ detune: [2400, -2400]
+ };
+
+ f.type = "peaking";
+ // Set starting points for all parameters of the filter. Start at 10 kHz for the center
+ // frequency, and the defaults for Q and gain.
+ f.frequency.setValueAtTime(parameters.freq[0], 0);
+ f.Q.setValueAtTime(parameters.Q[0], 0);
+ f.gain.setValueAtTime(parameters.gain[0], 0);
+ f.detune.setValueAtTime(parameters.detune[0], 0);
+
+ // Linear ramp each parameter
+ f.frequency.linearRampToValueAtTime(parameters.freq[1], renderDuration);
+ f.Q.linearRampToValueAtTime(parameters.Q[1], renderDuration);
+ f.gain.linearRampToValueAtTime(parameters.gain[1], renderDuration);
+ f.detune.linearRampToValueAtTime(parameters.detune[1], renderDuration);
+
+ context.startRendering()
+ .then(createFilterVerifier(createPeakingFilter, 3.3e-4, parameters, b.getChannelData(0),
+ "Output of peaking filter with automation of all parameters"));
+ }());
+
+ // Test that modulation of the frequency parameter of the filter works. A sinusoid of 440 Hz
+ // is the test signal that is applied to a bandpass biquad filter. The frequency parameter of
+ // the filter is modulated by a sinusoid at 103 Hz, and the frequency modulation varies from
+ // 116 to 412 Hz. (This test was taken from the description in
+ // https://github.com/WebAudio/web-audio-api/issues/509#issuecomment-94731355)
+ testPromises.push(function() {
+ var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+ // Create a graph with the sinusoidal source at 440 Hz as the input to a biquad filter.
+ var graph = configureGraph(context, 440);
+ var f = graph.filter;
+ var b = graph.source;
+
+ f.type = "bandpass";
+ f.Q.value = 5;
+ f.frequency.value = 264;
+
+ // Create the modulation source, a sinusoid with frequency 103 Hz and amplitude 148. (The
+ // amplitude of 148 is added to the filter's frequency value of 264 to produce a sinusoidal
+ // modulation of the frequency parameter from 116 to 412 Hz.)
+ var mod = context.createBufferSource();
+ var mbuffer = context.createBuffer(1, renderDuration * sampleRate, sampleRate);
+ var d = mbuffer.getChannelData(0);
+ var omega = 2 * Math.PI * 103 / sampleRate;
+ for (var k = 0; k < d.length; ++k) {
+ d[k] = 148 * Math.sin(omega * k);
+ }
+ mod.buffer = mbuffer;
+
+ mod.connect(f.frequency);
+
+ mod.start();
+ return context.startRendering()
+ .then(function (resultBuffer) {
+ var actual = resultBuffer.getChannelData(0);
+ // Compute the filter coefficients using the mod sine wave
+
+ var endFrame = Math.ceil(renderDuration * sampleRate);
+ var nCoef = endFrame;
+ var b0 = new Float64Array(nCoef);
+ var b1 = new Float64Array(nCoef);
+ var b2 = new Float64Array(nCoef);
+ var a1 = new Float64Array(nCoef);
+ var a2 = new Float64Array(nCoef);
+
+ // Generate the filter coefficients when the frequency varies from 116 to 248 Hz using
+ // the 103 Hz sinusoid.
+ for (var k = 0; k < nCoef; ++k) {
+ var freq = f.frequency.value + d[k];
+ var c = createBandpassFilter(freq / (sampleRate / 2), f.Q.value, f.gain.value);
+ b0[k] = c.b0;
+ b1[k] = c.b1;
+ b2[k] = c.b2;
+ a1[k] = c.a1;
+ a2[k] = c.a2;
+ }
+ reference = timeVaryingFilter(b.getChannelData(0),
+ {b0: b0, b1: b1, b2: b2, a1: a1, a2: a2});
+
+ compareChannels(actual, reference);
+ });
+ }());
+
+ // Wait for all tests
+ Promise.all(testPromises).then(function () {
+ SimpleTest.finish();
+ }, function () {
+ SimpleTest.finish();
+ });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeBandPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeBandPass.html
new file mode 100644
index 000000000..a3a1484f6
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeBandPass.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode Band Pass Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ // Create offline audio context.
+ var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+ // The filters we want to test.
+ var filterParameters = [{cutoff : 0, q : 0, gain : 1 },
+ {cutoff : 1, q : 0, gain : 1 },
+ {cutoff : 0.5, q : 0, gain : 1 },
+ {cutoff : 0.25, q : 1, gain : 1 },
+ ];
+
+ createTestAndRun(context, "bandpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeGetFrequencyResponse.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeGetFrequencyResponse.html
new file mode 100644
index 000000000..1576db1e8
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeGetFrequencyResponse.html
@@ -0,0 +1,261 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode All Pass Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+// Test the frequency response of a biquad filter. We compute the frequency response for a simple
+// peaking biquad filter and compare it with the expected frequency response. The actual filter
+// used doesn't matter since we're testing getFrequencyResponse and not the actual filter output.
+// The filters are extensively tested in other biquad tests.
+
+var context;
+
+// The biquad filter node.
+var filter;
+
+// The magnitude response of the biquad filter.
+var magResponse;
+
+// The phase response of the biquad filter.
+var phaseResponse;
+
+// Number of frequency samples to take.
+var numberOfFrequencies = 1000;
+
+// The filter parameters.
+var filterCutoff = 1000; // Hz.
+var filterQ = 1;
+var filterGain = 5; // Decibels.
+
+// The maximum allowed error in the magnitude response.
+var maxAllowedMagError = 5.7e-7;
+
+// The maximum allowed error in the phase response.
+var maxAllowedPhaseError = 4.7e-8;
+
+// The magnitudes and phases of the reference frequency response.
+var magResponse;
+var phaseResponse;
+
+// The magnitudes and phases of the reference frequency response.
+var expectedMagnitudes;
+var expectedPhases;
+
+// Convert frequency in Hz to a normalized frequency between 0 to 1 with 1 corresponding to the
+// Nyquist frequency.
+function normalizedFrequency(freqHz, sampleRate)
+{
+ var nyquist = sampleRate / 2;
+ return freqHz / nyquist;
+}
+
+// Get the filter response at a (normalized) frequency |f| for the filter with coefficients |coef|.
+function getResponseAt(coef, f)
+{
+ var b0 = coef.b0;
+ var b1 = coef.b1;
+ var b2 = coef.b2;
+ var a1 = coef.a1;
+ var a2 = coef.a2;
+
+ // H(z) = (b0 + b1 / z + b2 / z^2) / (1 + a1 / z + a2 / z^2)
+ //
+ // Compute H(exp(i * pi * f)). No native complex numbers in javascript, so break H(exp(i * pi * // f))
+ // in to the real and imaginary parts of the numerator and denominator. Let omega = pi * f.
+ // Then the numerator is
+ //
+ // b0 + b1 * cos(omega) + b2 * cos(2 * omega) - i * (b1 * sin(omega) + b2 * sin(2 * omega))
+ //
+ // and the denominator is
+ //
+ // 1 + a1 * cos(omega) + a2 * cos(2 * omega) - i * (a1 * sin(omega) + a2 * sin(2 * omega))
+ //
+ // Compute the magnitude and phase from the real and imaginary parts.
+
+ var omega = Math.PI * f;
+ var numeratorReal = b0 + b1 * Math.cos(omega) + b2 * Math.cos(2 * omega);
+ var numeratorImag = -(b1 * Math.sin(omega) + b2 * Math.sin(2 * omega));
+ var denominatorReal = 1 + a1 * Math.cos(omega) + a2 * Math.cos(2 * omega);
+ var denominatorImag = -(a1 * Math.sin(omega) + a2 * Math.sin(2 * omega));
+
+ var magnitude = Math.sqrt((numeratorReal * numeratorReal + numeratorImag * numeratorImag)
+ / (denominatorReal * denominatorReal + denominatorImag * denominatorImag));
+ var phase = Math.atan2(numeratorImag, numeratorReal) - Math.atan2(denominatorImag, denominatorReal);
+
+ if (phase >= Math.PI) {
+ phase -= 2 * Math.PI;
+ } else if (phase <= -Math.PI) {
+ phase += 2 * Math.PI;
+ }
+
+ return {magnitude : magnitude, phase : phase};
+}
+
+// Compute the reference frequency response for the biquad filter |filter| at the frequency samples
+// given by |frequencies|.
+function frequencyResponseReference(filter, frequencies)
+{
+ var sampleRate = filter.context.sampleRate;
+ var normalizedFreq = normalizedFrequency(filter.frequency.value, sampleRate);
+ var filterCoefficients = createFilter(filter.type, normalizedFreq, filter.Q.value, filter.gain.value);
+
+ var magnitudes = [];
+ var phases = [];
+
+ for (var k = 0; k < frequencies.length; ++k) {
+ var response = getResponseAt(filterCoefficients, normalizedFrequency(frequencies[k], sampleRate));
+ magnitudes.push(response.magnitude);
+ phases.push(response.phase);
+ }
+
+ return {magnitudes : magnitudes, phases : phases};
+}
+
+// Compute a set of linearly spaced frequencies.
+function createFrequencies(nFrequencies, sampleRate)
+{
+ var frequencies = new Float32Array(nFrequencies);
+ var nyquist = sampleRate / 2;
+ var freqDelta = nyquist / nFrequencies;
+
+ for (var k = 0; k < nFrequencies; ++k) {
+ frequencies[k] = k * freqDelta;
+ }
+
+ return frequencies;
+}
+
+function linearToDecibels(x)
+{
+ if (x) {
+ return 20 * Math.log(x) / Math.LN10;
+ } else {
+ return -1000;
+ }
+}
+
+// Look through the array and find any NaN or infinity. Returns the index of the first occurence or
+// -1 if none.
+function findBadNumber(signal)
+{
+ for (var k = 0; k < signal.length; ++k) {
+ if (!isValidNumber(signal[k])) {
+ return k;
+ }
+ }
+ return -1;
+}
+
+// Compute absolute value of the difference between phase angles, taking into account the wrapping
+// of phases.
+function absolutePhaseDifference(x, y)
+{
+ var diff = Math.abs(x - y);
+
+ if (diff > Math.PI) {
+ diff = 2 * Math.PI - diff;
+ }
+ return diff;
+}
+
+// Compare the frequency response with our expected response.
+function compareResponses(filter, frequencies, magResponse, phaseResponse)
+{
+ var expectedResponse = frequencyResponseReference(filter, frequencies);
+
+ expectedMagnitudes = expectedResponse.magnitudes;
+ expectedPhases = expectedResponse.phases;
+
+ var n = magResponse.length;
+ var success = true;
+ var badResponse = false;
+
+ var maxMagError = -1;
+ var maxMagErrorIndex = -1;
+
+ var k;
+ var hasBadNumber;
+
+ hasBadNumber = findBadNumber(magResponse);
+ ok (hasBadNumber < 0, "Magnitude response has NaN or infinity at " + hasBadNumber);
+
+ hasBadNumber = findBadNumber(phaseResponse);
+ ok (hasBadNumber < 0, "Phase response has NaN or infinity at " + hasBadNumber);
+
+ // These aren't testing the implementation itself. Instead, these are sanity checks on the
+ // reference. Failure here does not imply an error in the implementation.
+ hasBadNumber = findBadNumber(expectedMagnitudes);
+ ok (hasBadNumber < 0, "Expected magnitude response has NaN or infinity at " + hasBadNumber);
+
+ hasBadNumber = findBadNumber(expectedPhases);
+ ok (hasBadNumber < 0, "Expected phase response has NaN or infinity at " + hasBadNumber);
+
+ for (k = 0; k < n; ++k) {
+ var error = Math.abs(linearToDecibels(magResponse[k]) - linearToDecibels(expectedMagnitudes[k]));
+ if (error > maxMagError) {
+ maxMagError = error;
+ maxMagErrorIndex = k;
+ }
+ }
+
+ var message = "Magnitude error (" + maxMagError + " dB)";
+ message += " exceeded threshold at " + frequencies[maxMagErrorIndex];
+ message += " Hz. Actual: " + linearToDecibels(magResponse[maxMagErrorIndex]);
+ message += " dB, expected: " + linearToDecibels(expectedMagnitudes[maxMagErrorIndex]) + " dB.";
+ ok(maxMagError < maxAllowedMagError, message);
+
+ var maxPhaseError = -1;
+ var maxPhaseErrorIndex = -1;
+
+ for (k = 0; k < n; ++k) {
+ var error = absolutePhaseDifference(phaseResponse[k], expectedPhases[k]);
+ if (error > maxPhaseError) {
+ maxPhaseError = error;
+ maxPhaseErrorIndex = k;
+ }
+ }
+
+ message = "Phase error (radians) (" + maxPhaseError;
+ message += ") exceeded threshold at " + frequencies[maxPhaseErrorIndex];
+ message += " Hz. Actual: " + phaseResponse[maxPhaseErrorIndex];
+ message += " expected: " + expectedPhases[maxPhaseErrorIndex];
+
+ ok(maxPhaseError < maxAllowedPhaseError, message);
+}
+
+context = new AudioContext();
+
+filter = context.createBiquadFilter();
+
+// Arbitrarily test a peaking filter, but any kind of filter can be tested.
+filter.type = "peaking";
+filter.frequency.value = filterCutoff;
+filter.Q.value = filterQ;
+filter.gain.value = filterGain;
+
+var frequencies = createFrequencies(numberOfFrequencies, context.sampleRate);
+magResponse = new Float32Array(numberOfFrequencies);
+phaseResponse = new Float32Array(numberOfFrequencies);
+
+filter.getFrequencyResponse(frequencies, magResponse, phaseResponse);
+compareResponses(filter, frequencies, magResponse, phaseResponse);
+
+SimpleTest.finish();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeHighPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighPass.html
new file mode 100644
index 000000000..cb9aa274c
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighPass.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode High Pass Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ // Create offline audio context.
+ var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+ // The filters we want to test.
+ var filterParameters = [{cutoff : 0, q : 1, gain : 1 },
+ {cutoff : 1, q : 1, gain : 1 },
+ {cutoff : 0.25, q : 1, gain : 1 },
+ ];
+
+ createTestAndRun(context, "highpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeHighShelf.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighShelf.html
new file mode 100644
index 000000000..3581459b0
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighShelf.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode High Shelf Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ // Create offline audio context.
+ var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+ // The filters we want to test.
+ var filterParameters = [{cutoff : 0, q : 10, gain : 10 },
+ {cutoff : 1, q : 10, gain : 10 },
+ {cutoff : 0.25, q : 10, gain : 10 },
+ ];
+
+ createTestAndRun(context, "highshelf", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeLowPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowPass.html
new file mode 100644
index 000000000..b0c12558f
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowPass.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode Low Pass Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ // Create offline audio context.
+ var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+ // The filters we want to test.
+ var filterParameters = [{cutoff : 0, q : 1, gain : 1 },
+ {cutoff : 1, q : 1, gain : 1 },
+ {cutoff : 0.25, q : 1, gain : 1 },
+ {cutoff : 0.25, q : 1, gain : 1, detune : 100 },
+ {cutoff : 0.01, q : 1, gain : 1, detune : -200 },
+ ];
+ createTestAndRun(context, "lowpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeLowShelf.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowShelf.html
new file mode 100644
index 000000000..3c83bfaa3
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowShelf.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode Low Shelf Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+
+ // Create offline audio context.
+ var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+ // The filters we want to test.
+ var filterParameters = [{cutoff : 0, q : 10, gain : 10 },
+ {cutoff : 1, q : 10, gain : 10 },
+ {cutoff : 0.25, q : 10, gain : 10 },
+ ];
+
+ createTestAndRun(context, "lowshelf", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeNotch.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeNotch.html
new file mode 100644
index 000000000..551410c66
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeNotch.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode Notch Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ // Create offline audio context.
+ var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+ var filterParameters = [{cutoff : 0, q : 10, gain : 1 },
+ {cutoff : 1, q : 10, gain : 1 },
+ {cutoff : .5, q : 0, gain : 1 },
+ {cutoff : 0.25, q : 10, gain : 1 },
+ ];
+
+ createTestAndRun(context, "notch", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodePeaking.html b/dom/media/webaudio/test/blink/test_biquadFilterNodePeaking.html
new file mode 100644
index 000000000..33fcc225a
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodePeaking.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode Low Pass Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ // Create offline audio context.
+ var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+ // The filters we want to test.
+ var filterParameters = [{cutoff : 0, q : 10, gain : 10 },
+ {cutoff : 1, q : 10, gain : 10 },
+ {cutoff : .5, q : 0, gain : 10 },
+ {cutoff : 0.25, q : 10, gain : 10 },
+ ];
+
+ createTestAndRun(context, "peaking", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeTail.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeTail.html
new file mode 100644
index 000000000..fd02e734f
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeTail.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test BiquadFilterNode All Pass Filter</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="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ // A high sample rate shows the issue more clearly.
+ var sampleRate = 192000;
+ // Some short duration because we don't need to run the test for very long.
+ var testDurationSec = 0.5;
+ var testDurationFrames = testDurationSec * sampleRate;
+
+ // Amplitude experimentally determined to give a biquad output close to 1. (No attempt was
+ // made to produce exactly 1; it's not needed.)
+ var sourceAmplitude = 100;
+
+ // The output of the biquad filter should not change by more than this much between output
+ // samples. Threshold was determined experimentally.
+ var glitchThreshold = 0.01292;
+
+ // Test that a Biquad filter doesn't have it's output terminated because the input has gone
+ // away. Generally, when a source node is finished, it disconnects itself from any downstream
+ // nodes. This is the correct behavior. Nodes that have no inputs (disconnected) are
+ // generally assumed to output zeroes. This is also desired behavior. However, biquad
+ // filters have memory so they should not suddenly output zeroes when the input is
+ // disconnected. This test checks to see if the output doesn't suddenly change to zero.
+ var context = new OfflineAudioContext(1, testDurationFrames, sampleRate);
+
+ // Create an impulse source.
+ var buffer = context.createBuffer(1, 1, context.sampleRate);
+ buffer.getChannelData(0)[0] = sourceAmplitude;
+ var source = context.createBufferSource();
+ source.buffer = buffer;
+
+ // Create the biquad filter. It doesn't really matter what kind, so the default filter type
+ // and parameters is fine. Connect the source to it.
+ var biquad = context.createBiquadFilter();
+ source.connect(biquad);
+ biquad.connect(context.destination);
+
+ source.start();
+
+ context.startRendering().then(function(result) {
+ // There should be no large discontinuities in the output
+ var buffer = result.getChannelData(0);
+ var maxGlitchIndex = 0;
+ var maxGlitchValue = 0.0;
+ for (var i = 1; i < buffer.length; i++) {
+ var diff = Math.abs(buffer[i-1] - buffer[i]);
+ if (diff >= glitchThreshold) {
+ if (diff > maxGlitchValue) {
+ maxGlitchIndex = i;
+ maxGlitchValue = diff;
+ }
+ }
+ }
+ ok(maxGlitchIndex == 0, 'glitches detected in biquad output: maximum glitch at ' + maxGlitchIndex + ' with diff of ' + maxGlitchValue);
+ SimpleTest.finish();
+ })
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_iirFilterNode.html b/dom/media/webaudio/test/blink/test_iirFilterNode.html
new file mode 100644
index 000000000..47f936761
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_iirFilterNode.html
@@ -0,0 +1,467 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test IIRFilterNode GetFrequencyResponse</title>
+ <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+ <script type="text/javascript" src="webaudio.js"></script>
+ <script type="text/javascript" src="biquad-filters.js"></script>
+ <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ var sampleRate = 48000;
+ var testDurationSec = 1;
+ var testFrames = testDurationSec * sampleRate;
+
+ var testPromises = []
+ testPromises.push(function () {
+ // Test that the feedback coefficients are normalized. Do this be creating two
+ // IIRFilterNodes. One has normalized coefficients, and one doesn't. Compute the
+ // difference and make sure they're the same.
+ var context = new OfflineAudioContext(2, testFrames, sampleRate);
+
+ // Use a simple impulse as the source.
+ var buffer = context.createBuffer(1, 1, sampleRate);
+ buffer.getChannelData(0)[0] = 1;
+ var source = context.createBufferSource();
+ source.buffer = buffer;
+
+ // Gain node for computing the difference between the filters.
+ var gain = context.createGain();
+ gain.gain.value = -1;
+
+ // The IIR filters. Use a common feedforward array.
+ var ff = [1];
+
+ var fb1 = [1, .9];
+
+ var fb2 = new Float64Array(2);
+ // Scale the feedback coefficients by an arbitrary factor.
+ var coefScaleFactor = 2;
+ for (var k = 0; k < fb2.length; ++k) {
+ fb2[k] = coefScaleFactor * fb1[k];
+ }
+
+ var iir1 = context.createIIRFilter(ff, fb1);
+ var iir2 = context.createIIRFilter(ff, fb2);
+
+ // Create the graph. The output of iir1 (normalized coefficients) is channel 0, and the
+ // output of iir2 (unnormalized coefficients), with appropriate scaling, is channel 1.
+ var merger = context.createChannelMerger(2);
+ source.connect(iir1);
+ source.connect(iir2);
+ iir1.connect(merger, 0, 0);
+ iir2.connect(gain);
+
+ // The gain for the gain node should be set to compensate for the scaling of the
+ // coefficients. Since iir2 has scaled the coefficients by coefScaleFactor, the output is
+ // reduced by the same factor, so adjust the gain to scale the output of iir2 back up.
+ gain.gain.value = coefScaleFactor;
+ gain.connect(merger, 0, 1);
+
+ merger.connect(context.destination);
+
+ source.start();
+
+ // Rock and roll!
+
+ return context.startRendering().then(function (result) {
+ // Find the max amplitude of the result, which should be near zero.
+ var iir1Data = result.getChannelData(0);
+ var iir2Data = result.getChannelData(1);
+
+ // Threshold isn't exactly zero because the arithmetic is done differently between the
+ // IIRFilterNode and the BiquadFilterNode.
+ compareChannels(iir1Data, iir2Data);
+ });
+ }());
+
+ testPromises.push(function () {
+ // Create a simple 1-zero filter and compare with the expected output.
+ var context = new OfflineAudioContext(1, testFrames, sampleRate);
+
+ // Use a simple impulse as the source
+ var buffer = context.createBuffer(1, 1, sampleRate);
+ buffer.getChannelData(0)[0] = 1;
+ var source = context.createBufferSource();
+ source.buffer = buffer;
+
+ // The filter is y(n) = 0.5*(x(n) + x(n-1)), a simple 2-point moving average. This is
+ // rather arbitrary; keep it simple.
+
+ var iir = context.createIIRFilter([0.5, 0.5], [1]);
+
+ // Create the graph
+ source.connect(iir);
+ iir.connect(context.destination);
+
+ // Rock and roll!
+ source.start();
+
+ return context.startRendering().then(function (result) {
+ var actual = result.getChannelData(0);
+ var expected = new Float64Array(testFrames);
+ // The filter is a simple 2-point moving average of an impulse, so the first two values
+ // are non-zero and the rest are zero.
+ expected[0] = 0.5;
+ expected[1] = 0.5;
+ compareChannels(actual, expected);
+ });
+ }());
+
+ testPromises.push(function () {
+ // Create a simple 1-pole filter and compare with the expected output.
+
+ // The filter is y(n) + c*y(n-1)= x(n). The analytical response is (-c)^n, so choose a
+ // suitable number of frames to run the test for where the output isn't flushed to zero.
+ var c = 0.9;
+ var eps = 1e-20;
+ var duration = Math.floor(Math.log(eps) / Math.log(Math.abs(c)));
+ var context = new OfflineAudioContext(1, duration, sampleRate);
+
+ // Use a simple impulse as the source
+ var buffer = context.createBuffer(1, 1, sampleRate);
+ buffer.getChannelData(0)[0] = 1;
+ var source = context.createBufferSource();
+ source.buffer = buffer;
+
+ var iir = context.createIIRFilter([1], [1, c]);
+
+ // Create the graph
+ source.connect(iir);
+ iir.connect(context.destination);
+
+ // Rock and roll!
+ source.start();
+
+ return context.startRendering().then(function (result) {
+ var actual = result.getChannelData(0);
+ var expected = new Float64Array(actual.length);
+
+ // The filter is a simple 1-pole filter: y(n) = -c*y(n-k)+x(n), with an impulse as the
+ // input.
+ expected[0] = 1;
+ for (k = 1; k < testFrames; ++k) {
+ expected[k] = -c * expected[k-1];
+ }
+
+ compareChannels(actual, expected);
+ });
+ }());
+
+ // This function creates an IIRFilterNode equivalent to the specified
+ // BiquadFilterNode and compares the outputs. The
+ // outputs from the two filters should be virtually identical.
+ function testWithBiquadFilter(filterType) {
+ var context = new OfflineAudioContext(2, testFrames, sampleRate);
+
+ // Use a constant (step function) as the source
+ var buffer = context.createBuffer(1, testFrames, context.sampleRate);
+ for (var i = 0; i < testFrames; ++i) {
+ buffer.getChannelData(0)[i] = 1;
+ }
+ var source = context.createBufferSource();
+ source.buffer = buffer;
+
+ // Create the biquad. Choose some rather arbitrary values for Q and gain for the biquad
+ // so that the shelf filters aren't identical.
+ var biquad = context.createBiquadFilter();
+ biquad.type = filterType;
+ biquad.Q.value = 10;
+ biquad.gain.value = 10;
+
+ // Create the equivalent IIR Filter node by computing the coefficients of the given biquad
+ // filter type.
+ var nyquist = sampleRate / 2;
+ var coef = createFilter(filterType,
+ biquad.frequency.value / nyquist,
+ biquad.Q.value,
+ biquad.gain.value);
+
+ var iir = context.createIIRFilter([coef.b0, coef.b1, coef.b2], [1, coef.a1, coef.a2]);
+
+ var merger = context.createChannelMerger(2);
+ // Create the graph
+ source.connect(biquad);
+ source.connect(iir);
+
+ biquad.connect(merger, 0, 0);
+ iir.connect(merger, 0, 1);
+
+ merger.connect(context.destination);
+
+ // Rock and roll!
+ source.start();
+
+ return context.startRendering().then(function (result) {
+ // Find the max amplitude of the result, which should be near zero.
+ var expected = result.getChannelData(0);
+ var actual = result.getChannelData(1);
+ compareChannels(actual, expected);
+ });
+ }
+
+ biquadFilterTypes = ["lowpass", "highpass", "bandpass", "notch",
+ "allpass", "lowshelf", "highshelf", "peaking"];
+
+ // Create a set of tasks based on biquadTestConfigs.
+ for (var i = 0; i < biquadFilterTypes.length; ++i) {
+ testPromises.push(testWithBiquadFilter(biquadFilterTypes[i]));
+ }
+
+ testPromises.push(function () {
+ // Multi-channel test. Create a biquad filter and the equivalent IIR filter. Filter the
+ // same multichannel signal and compare the results.
+ var nChannels = 3;
+ var context = new OfflineAudioContext(nChannels, testFrames, sampleRate);
+
+ // Create a set of oscillators as the multi-channel source.
+ var source = [];
+
+ for (k = 0; k < nChannels; ++k) {
+ source[k] = context.createOscillator();
+ source[k].type = "sawtooth";
+ // The frequency of the oscillator is pretty arbitrary, but each oscillator should have a
+ // different frequency.
+ source[k].frequency.value = 100 + k * 100;
+ }
+
+ var merger = context.createChannelMerger(3);
+
+ var biquad = context.createBiquadFilter();
+
+ // Create the equivalent IIR Filter node.
+ var nyquist = sampleRate / 2;
+ var coef = createFilter(biquad.type,
+ biquad.frequency.value / nyquist,
+ biquad.Q.value,
+ biquad.gain.value);
+ var fb = [1, coef.a1, coef.a2];
+ var ff = [coef.b0, coef.b1, coef.b2];
+
+ var iir = context.createIIRFilter(ff, fb);
+ // Gain node to compute the difference between the IIR and biquad filter.
+ var gain = context.createGain();
+ gain.gain.value = -1;
+
+ // Create the graph.
+ for (k = 0; k < nChannels; ++k)
+ source[k].connect(merger, 0, k);
+
+ merger.connect(biquad);
+ merger.connect(iir);
+ iir.connect(gain);
+ biquad.connect(context.destination);
+ gain.connect(context.destination);
+
+ for (k = 0; k < nChannels; ++k)
+ source[k].start();
+
+ return context.startRendering().then(function (result) {
+ var errorThresholds = [3.7671e-5, 3.0071e-5, 2.6241e-5];
+
+ // Check the difference signal on each channel
+ for (channel = 0; channel < result.numberOfChannels; ++channel) {
+ // Find the max amplitude of the result, which should be near zero.
+ var data = result.getChannelData(channel);
+ var maxError = data.reduce(function(reducedValue, currentValue) {
+ return Math.max(reducedValue, Math.abs(currentValue));
+ });
+
+ ok(maxError <= errorThresholds[channel], "Max difference between IIR and Biquad on channel " + channel);
+ }
+ });
+ }());
+
+ testPromises.push(function () {
+ // Apply an IIRFilter to the given input signal.
+ //
+ // IIR filter in the time domain is
+ //
+ // y[n] = sum(ff[k]*x[n-k], k, 0, M) - sum(fb[k]*y[n-k], k, 1, N)
+ //
+ function iirFilter(input, feedforward, feedback) {
+ // For simplicity, create an x buffer that contains the input, and a y buffer that contains
+ // the output. Both of these buffers have an initial work space to implement the initial
+ // memory of the filter.
+ var workSize = Math.max(feedforward.length, feedback.length);
+ var x = new Float32Array(input.length + workSize);
+
+ // Float64 because we want to match the implementation that uses doubles to minimize
+ // roundoff.
+ var y = new Float64Array(input.length + workSize);
+
+ // Copy the input over.
+ for (var k = 0; k < input.length; ++k)
+ x[k + feedforward.length] = input[k];
+
+ // Run the filter
+ for (var n = 0; n < input.length; ++n) {
+ var index = n + workSize;
+ var yn = 0;
+ for (var k = 0; k < feedforward.length; ++k)
+ yn += feedforward[k] * x[index - k];
+ for (var k = 0; k < feedback.length; ++k)
+ yn -= feedback[k] * y[index - k];
+
+ y[index] = yn;
+ }
+
+ return y.slice(workSize).map(Math.fround);
+ }
+
+ // Cascade the two given biquad filters to create one IIR filter.
+ function cascadeBiquads(f1Coef, f2Coef) {
+ // The biquad filters are:
+ //
+ // f1 = (b10 + b11/z + b12/z^2)/(1 + a11/z + a12/z^2);
+ // f2 = (b20 + b21/z + b22/z^2)/(1 + a21/z + a22/z^2);
+ //
+ // To cascade them, multiply the two transforms together to get a fourth order IIR filter.
+
+ var numProduct = [f1Coef.b0 * f2Coef.b0,
+ f1Coef.b0 * f2Coef.b1 + f1Coef.b1 * f2Coef.b0,
+ f1Coef.b0 * f2Coef.b2 + f1Coef.b1 * f2Coef.b1 + f1Coef.b2 * f2Coef.b0,
+ f1Coef.b1 * f2Coef.b2 + f1Coef.b2 * f2Coef.b1,
+ f1Coef.b2 * f2Coef.b2
+ ];
+
+ var denProduct = [1,
+ f2Coef.a1 + f1Coef.a1,
+ f2Coef.a2 + f1Coef.a1 * f2Coef.a1 + f1Coef.a2,
+ f1Coef.a1 * f2Coef.a2 + f1Coef.a2 * f2Coef.a1,
+ f1Coef.a2 * f2Coef.a2
+ ];
+
+ return {
+ ff: numProduct,
+ fb: denProduct
+ }
+ }
+
+ // Find the magnitude of the root of the quadratic that has the maximum magnitude.
+ //
+ // The quadratic is z^2 + a1 * z + a2 and we want the root z that has the largest magnitude.
+ function largestRootMagnitude(a1, a2) {
+ var discriminant = a1 * a1 - 4 * a2;
+ if (discriminant < 0) {
+ // Complex roots: -a1/2 +/- i*sqrt(-d)/2. Thus the magnitude of each root is the same
+ // and is sqrt(a1^2/4 + |d|/4)
+ var d = Math.sqrt(-discriminant);
+ return Math.hypot(a1 / 2, d / 2);
+ } else {
+ // Real roots
+ var d = Math.sqrt(discriminant);
+ return Math.max(Math.abs((-a1 + d) / 2), Math.abs((-a1 - d) / 2));
+ }
+ }
+
+ // Cascade 2 lowpass biquad filters and compare that with the equivalent 4th order IIR
+ // filter.
+
+ var nyquist = sampleRate / 2;
+ // Compute the coefficients of a lowpass filter.
+
+ // First some preliminary stuff. Compute the coefficients of the biquad. This is used to
+ // figure out how frames to use in the test.
+ var biquadType = "lowpass";
+ var biquadCutoff = 350;
+ var biquadQ = 5;
+ var biquadGain = 1;
+
+ var coef = createFilter(biquadType,
+ biquadCutoff / nyquist,
+ biquadQ,
+ biquadGain);
+
+ // Cascade the biquads together to create an equivalent IIR filter.
+ var cascade = cascadeBiquads(coef, coef);
+
+ // Since we're cascading two identical biquads, the root of denominator of the IIR filter is
+ // repeated, so the root of the denominator with the largest magnitude occurs twice. The
+ // impulse response of the IIR filter will be roughly c*(r*r)^n at time n, where r is the
+ // root of largest magnitude. This approximation gets better as n increases. We can use
+ // this to get a rough idea of when the response has died down to a small value.
+
+ // This is the value we will use to determine how many frames to render. Rendering too many
+ // is a waste of time and also makes it hard to compare the actual result to the expected
+ // because the magnitudes are so small that they could be mostly round-off noise.
+ //
+ // Find magnitude of the root with largest magnitude
+ var rootMagnitude = largestRootMagnitude(coef.a1, coef.a2);
+
+ // Find n such that |r|^(2*n) <= eps. That is, n = log(eps)/(2*log(r)). Somewhat
+ // arbitrarily choose eps = 1e-20;
+ var eps = 1e-20;
+ var framesForTest = Math.floor(Math.log(eps) / (2 * Math.log(rootMagnitude)));
+
+ // We're ready to create the graph for the test. The offline context has two channels:
+ // channel 0 is the expected (cascaded biquad) result and channel 1 is the actual IIR filter
+ // result.
+ var context = new OfflineAudioContext(2, framesForTest, sampleRate);
+
+ // Use a simple impulse with a large (arbitrary) amplitude as the source
+ var amplitude = 1;
+ var buffer = context.createBuffer(1, testFrames, sampleRate);
+ buffer.getChannelData(0)[0] = amplitude;
+ var source = context.createBufferSource();
+ source.buffer = buffer;
+
+ // Create the two biquad filters. Doesn't really matter what, but for simplicity we choose
+ // identical lowpass filters with the same parameters.
+ var biquad1 = context.createBiquadFilter();
+ biquad1.type = biquadType;
+ biquad1.frequency.value = biquadCutoff;
+ biquad1.Q.value = biquadQ;
+
+ var biquad2 = context.createBiquadFilter();
+ biquad2.type = biquadType;
+ biquad2.frequency.value = biquadCutoff;
+ biquad2.Q.value = biquadQ;
+
+ var iir = context.createIIRFilter(cascade.ff, cascade.fb);
+
+ // Create the merger to get the signals into multiple channels
+ var merger = context.createChannelMerger(2);
+
+ // Create the graph, filtering the source through two biquads.
+ source.connect(biquad1);
+ biquad1.connect(biquad2);
+ biquad2.connect(merger, 0, 0);
+
+ source.connect(iir);
+ iir.connect(merger, 0, 1);
+
+ merger.connect(context.destination);
+
+ // Now filter the source through the IIR filter.
+ var y = iirFilter(buffer.getChannelData(0), cascade.ff, cascade.fb);
+
+ // Rock and roll!
+ source.start();
+
+ return context.startRendering().then(function(result) {
+ var expected = result.getChannelData(0);
+ var actual = result.getChannelData(1);
+
+ compareChannels(actual, expected);
+
+ });
+ }());
+
+ // Wait for all tests
+ Promise.all(testPromises).then(function () {
+ SimpleTest.finish();
+ }, function () {
+ SimpleTest.finish();
+ });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_iirFilterNodeGetFrequencyResponse.html b/dom/media/webaudio/test/blink/test_iirFilterNodeGetFrequencyResponse.html
new file mode 100644
index 000000000..cb5cf33ed
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_iirFilterNodeGetFrequencyResponse.html
@@ -0,0 +1,97 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+ <title>Test IIRFilterNode GetFrequencyResponse</title>
+ <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+ <script type="text/javascript" src="webaudio.js"></script>
+ <script type="text/javascript" src="biquad-filters.js"></script>
+ <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+ // Modified from WebKit/LayoutTests/webaudio/iirfilter-getFrequencyResponse.html
+ var sampleRate = 48000;
+ var testDurationSec = 0.01;
+
+ // Compute a set of linearly spaced frequencies.
+ function createFrequencies(nFrequencies, sampleRate)
+ {
+ var frequencies = new Float32Array(nFrequencies);
+ var nyquist = sampleRate / 2;
+ var freqDelta = nyquist / nFrequencies;
+
+ for (var k = 0; k < nFrequencies; ++k) {
+ frequencies[k] = k * freqDelta;
+ }
+
+ return frequencies;
+ }
+
+ // Number of frequency samples to take.
+ var numberOfFrequencies = 1000;
+
+ var context = new OfflineAudioContext(1, testDurationSec * sampleRate, sampleRate);
+
+ var frequencies = createFrequencies(numberOfFrequencies, context.sampleRate);
+
+ // 1-Pole IIR Filter
+ var iir = context.createIIRFilter([1], [1, -0.9]);
+
+ var iirMag = new Float32Array(numberOfFrequencies);
+ var iirPhase = new Float32Array(numberOfFrequencies);
+ var trueMag = new Float32Array(numberOfFrequencies);
+ var truePhase = new Float32Array(numberOfFrequencies);
+
+ // The IIR filter is
+ // H(z) = 1/(1 - 0.9*z^(-1)).
+ //
+ // The frequency response is
+ // H(exp(j*w)) = 1/(1 - 0.9*exp(-j*w)).
+ //
+ // Thus, the magnitude is
+ // |H(exp(j*w))| = 1/sqrt(1.81-1.8*cos(w)).
+ //
+ // The phase is
+ // arg(H(exp(j*w)) = atan(0.9*sin(w)/(.9*cos(w)-1))
+
+ var frequencyScale = Math.PI / (sampleRate / 2);
+
+ for (var k = 0; k < frequencies.length; ++k) {
+ var omega = frequencyScale * frequencies[k];
+ trueMag[k] = 1/Math.sqrt(1.81-1.8*Math.cos(omega));
+ truePhase[k] = Math.atan(0.9 * Math.sin(omega) / (0.9 * Math.cos(omega) - 1));
+ }
+
+ iir.getFrequencyResponse(frequencies, iirMag, iirPhase);
+ compareChannels(iirMag, trueMag);
+ compareChannels(iirPhase, truePhase);
+
+ // Compare IIR and Biquad Filter
+ // Create an IIR filter equivalent to the biquad filter. Compute the frequency response for both and verify that they are the same.
+ var biquad = context.createBiquadFilter();
+ var coef = createFilter(biquad.type,
+ biquad.frequency.value / (context.sampleRate / 2),
+ biquad.Q.value,
+ biquad.gain.value);
+
+ var iir = context.createIIRFilter([coef.b0, coef.b1, coef.b2], [1, coef.a1, coef.a2]);
+
+ var biquadMag = new Float32Array(numberOfFrequencies);
+ var biquadPhase = new Float32Array(numberOfFrequencies);
+ var iirMag = new Float32Array(numberOfFrequencies);
+ var iirPhase = new Float32Array(numberOfFrequencies);
+
+ biquad.getFrequencyResponse(frequencies, biquadMag, biquadPhase);
+ iir.getFrequencyResponse(frequencies, iirMag, iirPhase);
+ compareChannels(iirMag, biquadMag);
+ compareChannels(iirPhase, biquadPhase);
+
+ SimpleTest.finish();
+});
+</script>
+</pre>
+</body>
+</html>