Add support for WaveShaperNode.oversample

Source/core/platform/audio/DownSampler.cpp

Index: Source/core/platform/audio/DownSampler.cpp
diff --git a/Source/core/platform/audio/DownSampler.cpp b/Source/core/platform/audio/DownSampler.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..3db140f3694c13709e89e68c087ce6d40ca2cb14
--- /dev/null
+++ b/Source/core/platform/audio/DownSampler.cpp
@@ -0,0 +1,151 @@
+/*
+ * Copyright (C) 2013 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ *
+ *     * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *     * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following disclaimer
+ * in the documentation and/or other materials provided with the
+ * distribution.
+ *     * Neither the name of Google Inc. nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+
+#if ENABLE(WEB_AUDIO)
+
+#include "core/platform/audio/DownSampler.h"
+
+#include "wtf/MathExtras.h"
+
+namespace WebCore {
+
+DownSampler::DownSampler(size_t inputBlockSize)
+    : m_inputBlockSize(inputBlockSize)
+    , m_reducedKernel(DefaultKernelSize / 2)
+    , m_tempBuffer(inputBlockSize / 2)
+    , m_inputBuffer(inputBlockSize * 2)
+    , m_convolver(inputBlockSize / 2) // runs at 1/2 source sample-rate
+{
+    initializeKernel();
+}
+
+void DownSampler::initializeKernel()
+{
+    // Blackman window parameters.
+    double alpha = 0.16;
+    double a0 = 0.5 * (1.0 - alpha);
+    double a1 = 0.5;
+    double a2 = 0.5 * alpha;
+
+    int n = DefaultKernelSize;
+    int halfSize = n / 2;
+
+    // Half-band filter.
+    double sincScaleFactor = 0.5;
+
+    // Compute only the odd terms because the even ones are zero, except
+    // right in the middle at halfSize, which is 0.5 and we'll handle specially during processing
+    // after doing the main convolution using m_reducedKernel.
+    for (int i = 1; i < n; i += 2) {
+        // Compute the sinc() with offset.
+        double s = sincScaleFactor * piDouble * (i - halfSize);
+        double sinc = !s ? 1.0 : sin(s) / s;
+        sinc *= sincScaleFactor;
+
+        // Compute Blackman window, matching the offset of the sinc().
+        double x = static_cast<double>(i) / n;
+        double window = a0 - a1 * cos(2.0 * piDouble * x) + a2 * cos(4.0 * piDouble * x);
+
+        // Window the sinc() function.
+        // Then store only the odd terms in the kernel.
+        // In a sense, this is shifting forward in time by one sample-frame at the destination sample-rate.
+        m_reducedKernel[(i - 1) / 2] = sinc * window;
+    }
+}
+
+void DownSampler::process(const float* sourceP, float* destP, size_t sourceFramesToProcess)
+{
+    bool isInputBlockSizeGood = sourceFramesToProcess == m_inputBlockSize;
+    ASSERT(isInputBlockSizeGood);
+    if (!isInputBlockSizeGood)
+        return;
+
+    size_t destFramesToProcess = sourceFramesToProcess / 2;
+
+    bool isTempBufferGood = destFramesToProcess == m_tempBuffer.size();
+    ASSERT(isTempBufferGood);
+    if (!isTempBufferGood)
+        return;
+
+    bool isReducedKernelGood = m_reducedKernel.size() == DefaultKernelSize / 2;
+    ASSERT(isReducedKernelGood);
+    if (!isReducedKernelGood)
+        return;
+
+    size_t halfSize = DefaultKernelSize / 2;
+
+    // Copy source samples to 2nd half of input buffer.
+    bool isInputBufferGood = m_inputBuffer.size() == sourceFramesToProcess * 2 && halfSize <= sourceFramesToProcess;
+    ASSERT(isInputBufferGood);
+    if (!isInputBufferGood)
+        return;
+
+    float* inputP = m_inputBuffer.data() + sourceFramesToProcess;
+    memcpy(inputP, sourceP, sizeof(float) * sourceFramesToProcess);
+
+    // Copy the odd sample-frames from sourceP, delayed by one sample-frame (destination sample-rate)
+    // to match shifting forward in time in m_reducedKernel.
+    float* oddSamplesP = m_tempBuffer.data();
+    for (unsigned i = 0; i < destFramesToProcess; ++i)
+        oddSamplesP[i] = inputP[i * 2 - 1];
+
+    // Actually process oddSamplesP with m_reducedKernel for efficiency.
+    // The theoretical kernel is double this size with 0 values for even terms (except center).
+    m_convolver.process(&m_reducedKernel, oddSamplesP, destP, destFramesToProcess);
+
+    // Now, account for the 0.5 term right in the middle of the kernel.
+    // This amounts to a delay-line of length halfSize (at the source sample-rate),
+    // scaled by 0.5.
+
+    // Sum into the destination.
+    for (unsigned i = 0; i < destFramesToProcess; ++i)
+        destP[i] += 0.5 * inputP[i * 2 - halfSize];
+
+    // Copy 2nd half of input buffer to 1st half.
+    memcpy(m_inputBuffer.data(), inputP, sizeof(float) * sourceFramesToProcess);
+}
+
+void DownSampler::reset()
+{
+    m_convolver.reset();
+    m_inputBuffer.zero();
+}
+
+size_t DownSampler::latencyFrames() const
+{
+    // Divide by two since this is a linear phase kernel and the delay is at the center of the kernel.
+    return m_reducedKernel.size() / 2;
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_AUDIO)