Merge Webkit at r70949: Initial merge by git.

Change-Id: I77b8645c083b5d0da8dba73ed01d4014aab9848e

1 files changed, 143 insertions, 0 deletions

diff --git a/WebCore/platform/audio/AudioResamplerKernel.cpp b/WebCore/platform/audio/AudioResamplerKernel.cpp
new file mode 100644
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--- /dev/null
+++ b/WebCore/platform/audio/AudioResamplerKernel.cpp
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+/*
+ * Copyright (C) 2010, 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:
+ * 1.  Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2.  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.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS 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 APPLE INC. OR ITS 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 "AudioResamplerKernel.h"
+
+#include "AudioResampler.h"
+#include <algorithm>
+
+using namespace std;
+
+namespace WebCore {
+    
+const size_t AudioResamplerKernel::MaxFramesToProcess = 128;
+
+AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler)
+    : m_resampler(resampler)
+    // The buffer size must be large enough to hold up to two extra sample frames for the linear interpolation.
+    , m_sourceBuffer(2 + static_cast<int>(MaxFramesToProcess * AudioResampler::MaxRate))
+    , m_virtualReadIndex(0.0)
+    , m_fillIndex(0)
+{
+    m_lastValues[0] = 0.0f;
+    m_lastValues[1] = 0.0f;
+}
+
+float* AudioResamplerKernel::getSourcePointer(size_t framesToProcess, size_t* numberOfSourceFramesNeededP)
+{
+    ASSERT(framesToProcess <= MaxFramesToProcess);
+    
+    // Calculate the next "virtual" index.  After process() is called, m_virtualReadIndex will equal this value.
+    double nextFractionalIndex = m_virtualReadIndex + framesToProcess * rate();
+
+    // Because we're linearly interpolating between the previous and next sample we need to round up so we include the next sample.
+    int endIndex = static_cast<int>(nextFractionalIndex + 1.0); // round up to next integer index
+
+    // Determine how many input frames we'll need.
+    // We need to fill the buffer up to and including endIndex (so add 1) but we've already buffered m_fillIndex frames from last time.
+    size_t framesNeeded = 1 + endIndex - m_fillIndex;
+    if (numberOfSourceFramesNeededP)
+        *numberOfSourceFramesNeededP = framesNeeded;
+
+    // Do bounds checking for the source buffer.
+    bool isGood = m_fillIndex < m_sourceBuffer.size() && m_fillIndex + framesNeeded <= m_sourceBuffer.size();
+    ASSERT(isGood);
+    if (!isGood)
+        return 0;
+
+    return m_sourceBuffer.data() + m_fillIndex;
+}
+
+void AudioResamplerKernel::process(float* destination, size_t framesToProcess)
+{
+    ASSERT(framesToProcess <= MaxFramesToProcess);
+
+    float* source = m_sourceBuffer.data();
+    
+    double rate = this->rate();
+    rate = max(0.0, rate);
+    rate = min(AudioResampler::MaxRate, rate);
+    
+    // Start out with the previous saved values (if any).
+    if (m_fillIndex > 0) {
+        source[0] = m_lastValues[0];
+        source[1] = m_lastValues[1];
+    }
+
+    // Make a local copy.
+    double virtualReadIndex = m_virtualReadIndex;
+    
+    // Sanity check source buffer access.
+    ASSERT(framesToProcess > 0);
+    ASSERT(virtualReadIndex >= 0 && 1 + static_cast<unsigned>(virtualReadIndex + (framesToProcess - 1) * rate) < m_sourceBuffer.size());
+
+    // Do the linear interpolation.
+    int n = framesToProcess;
+    while (n--) {
+        unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
+        double interpolationFactor = virtualReadIndex - readIndex;
+
+        double sample1 = source[readIndex];
+        double sample2 = source[readIndex + 1];
+
+        double sample = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;
+
+        *destination++ = static_cast<float>(sample);
+
+        virtualReadIndex += rate;
+    }                        
+
+    // Save the last two sample-frames which will later be used at the beginning of the source buffer the next time around.
+    int readIndex = static_cast<int>(virtualReadIndex);
+    m_lastValues[0] = source[readIndex];
+    m_lastValues[1] = source[readIndex + 1];
+    m_fillIndex = 2;
+
+    // Wrap the virtual read index back to the start of the buffer.
+    virtualReadIndex -= readIndex;
+
+    // Put local copy back into member variable.
+    m_virtualReadIndex = virtualReadIndex;
+}
+
+void AudioResamplerKernel::reset()
+{
+    m_virtualReadIndex = 0.0;
+    m_fillIndex = 0;
+    m_lastValues[0] = 0.0f;
+    m_lastValues[1] = 0.0f;
+}
+
+double AudioResamplerKernel::rate() const
+{
+    return m_resampler->rate();
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_AUDIO)