Add m-esr52 at 52.6.0

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>