Implement IIRFilter node for WebAudio.

third_party/WebKit/Source/modules/webaudio/IIRProcessor.cpp

Index: third_party/WebKit/Source/modules/webaudio/IIRProcessor.cpp
diff --git a/third_party/WebKit/Source/modules/webaudio/IIRProcessor.cpp b/third_party/WebKit/Source/modules/webaudio/IIRProcessor.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..5bb50753e4a584c558e409b18c903839151dd1a6
--- /dev/null
+++ b/third_party/WebKit/Source/modules/webaudio/IIRProcessor.cpp
@@ -0,0 +1,81 @@
+// Copyright 2015 The Chromium Authors. All rights reserved.

[tkent, 2016/01/13 03:28:28]

2016

[Raymond Toy, 2016/01/13 18:30:53]

Done.

+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "modules/webaudio/IIRProcessor.h"
+
+#include "modules/webaudio/IIRDSPKernel.h"
+
+namespace blink {
+
+IIRProcessor::IIRProcessor(float sampleRate, size_t numberOfChannels, const Vector<double>& feedforwardCoef, const Vector<double>& feedbackCoef)
+    : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
+{
+    unsigned feedbackLength = feedbackCoef.size();
+    unsigned feedforwardLength = feedforwardCoef.size();
+    ASSERT(feedbackLength > 0);
+    ASSERT(feedforwardLength > 0);
+
+    m_feedforward.allocate(feedforwardLength);
+    m_feedback.allocate(feedbackLength);
+    m_feedforward.copyToRange(feedforwardCoef.data(), 0, feedforwardLength);
+    m_feedback.copyToRange(feedbackCoef.data(), 0, feedbackLength);
+
+    // Need to scale the feedback and feedforward coefficients appropriately. (It's up to the caller
+    // to ensure feedbackCoef[0] is not 0!)
+    ASSERT(feedbackCoef[0]);

[tkent, 2016/01/13 03:28:28]

nit: You may compare with 0.  ASSERT(feedbackCoef[0] != 0);

[Raymond Toy, 2016/01/13 18:30:53]

Done.

+
+    if (feedbackCoef[0] != 1) {
+        // The provided filter is:
+        //
+        //   a[0]*y(n) + a[1]*y(n-1) + ... = b[0]*x(n) + b[1]*x(n-1) + ...
+        //
+        // We want the leading coefficient of y(n) to be 1:
+        //
+        //   y(n) + a[1]/a[0]*y(n-1) + ... = b[0]/a[0]*x(n) + b[1]/a[0]*x(n-1) + ...
+        //
+        // Thus, the feedback and feedforward coefficients need to be scaled by 1/a[0].
+        float scale = feedbackCoef[0];
+        for (unsigned k = 1; k < feedbackLength; ++k)
+            m_feedback[k] /= scale;
+
+        for (unsigned k = 0; k < feedforwardLength; ++k)
+            m_feedforward[k] /= scale;
+
+        // The IIRFilter checks to make sure this coefficient is 1, so make it so.
+        m_feedback[0] = 1;
+    }
+
+    m_responseKernel = adoptPtr(new IIRDSPKernel(this));
+}
+
+IIRProcessor::~IIRProcessor()
+{
+    if (isInitialized())
+        uninitialize();
+}
+
+PassOwnPtr<AudioDSPKernel> IIRProcessor::createKernel()
+{
+    return adoptPtr(new IIRDSPKernel(this));
+}
+
+void IIRProcessor::process(const AudioBus* source, AudioBus* destination, size_t framesToProcess)
+{
+    if (!isInitialized()) {
+        destination->zero();
+        return;
+    }
+
+    // For each channel of our input, process using the corresponding IIRDSPKernel into the output
+    // channel.
+    for (unsigned i = 0; i < m_kernels.size(); ++i)
+        m_kernels[i]->process(source->channel(i)->data(), destination->channel(i)->mutableData(), framesToProcess);
+}
+
+void IIRProcessor::getFrequencyResponse(int nFrequencies, const float* frequencyHz, float* magResponse, float* phaseResponse)
+{
+    m_responseKernel->getFrequencyResponse(nFrequencies, frequencyHz, magResponse, phaseResponse);
+}
+
+} // blink