Merge WebKit at r71558: Initial merge by git.

Change-Id: Ib345578fa29df7e4bc72b4f00e4a6fddcb754c4c

1 files changed, 300 insertions, 0 deletions

diff --git a/WebCore/webaudio/RealtimeAnalyser.cpp b/WebCore/webaudio/RealtimeAnalyser.cpp
new file mode 100644
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+++ b/WebCore/webaudio/RealtimeAnalyser.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 "RealtimeAnalyser.h"
+
+#include "AudioBus.h"
+#include "AudioUtilities.h"
+#include "FFTFrame.h"
+
+#if ENABLE(3D_CANVAS)
+#include "Float32Array.h"
+#include "Uint8Array.h"
+#endif
+
+#include <algorithm>
+#include <limits.h>
+#include <wtf/Complex.h>
+#include <wtf/Threading.h>
+
+using namespace std;
+
+namespace WebCore {
+
+const double RealtimeAnalyser::DefaultSmoothingTimeConstant  = 0.8;
+const double RealtimeAnalyser::DefaultMinDecibels = -100.0;
+const double RealtimeAnalyser::DefaultMaxDecibels = -30.0;
+
+const unsigned RealtimeAnalyser::DefaultFFTSize = 2048;
+const unsigned RealtimeAnalyser::MaxFFTSize = 2048;
+const unsigned RealtimeAnalyser::InputBufferSize = RealtimeAnalyser::MaxFFTSize * 2;
+
+RealtimeAnalyser::RealtimeAnalyser()
+    : m_inputBuffer(InputBufferSize)
+    , m_writeIndex(0)
+    , m_fftSize(DefaultFFTSize)
+    , m_magnitudeBuffer(DefaultFFTSize / 2)
+    , m_smoothingTimeConstant(DefaultSmoothingTimeConstant)
+    , m_minDecibels(DefaultMinDecibels)
+    , m_maxDecibels(DefaultMaxDecibels)
+{
+    m_analysisFrame = adoptPtr(new FFTFrame(DefaultFFTSize));
+}
+
+RealtimeAnalyser::~RealtimeAnalyser()
+{
+}
+
+void RealtimeAnalyser::reset()
+{
+    m_writeIndex = 0;
+    m_inputBuffer.zero();
+    m_magnitudeBuffer.zero();
+}
+
+void RealtimeAnalyser::setFftSize(size_t size)
+{
+    ASSERT(isMainThread());
+
+    // Only allow powers of two.
+    unsigned log2size = static_cast<unsigned>(log2(size));
+    bool isPOT(1UL << log2size == size);
+    
+    if (!isPOT || size > MaxFFTSize) {
+        // FIXME: It would be good to also set an exception.
+        return;
+    }
+
+    if (m_fftSize != size) {
+        m_analysisFrame = adoptPtr(new FFTFrame(m_fftSize));
+        m_magnitudeBuffer.resize(size);
+        m_fftSize = size;
+    }
+}
+
+void RealtimeAnalyser::writeInput(AudioBus* bus, size_t framesToProcess)
+{
+    bool isBusGood = bus && bus->numberOfChannels() > 0 && bus->channel(0)->length() >= framesToProcess;
+    ASSERT(isBusGood);
+    if (!isBusGood)
+        return;
+        
+    // FIXME : allow to work with non-FFTSize divisible chunking
+    bool isDestinationGood = m_writeIndex < m_inputBuffer.size() && m_writeIndex + framesToProcess <= m_inputBuffer.size();
+    ASSERT(isDestinationGood);
+    if (!isDestinationGood)
+        return;    
+    
+    // Perform real-time analysis
+    // FIXME : for now just use left channel (must mix if stereo source)
+    float* source = bus->channel(0)->data();
+
+    // The source has already been sanity checked with isBusGood above.
+    
+    memcpy(m_inputBuffer.data() + m_writeIndex, source, sizeof(float) * framesToProcess);
+
+    m_writeIndex += framesToProcess;
+    if (m_writeIndex >= InputBufferSize)
+        m_writeIndex = 0;
+}
+
+namespace {
+
+void applyWindow(float* p, size_t n)
+{
+    ASSERT(isMainThread());
+    
+    // Blackman window
+    double alpha = 0.16;
+    double a0 = 0.5 * (1.0 - alpha);
+    double a1 = 0.5;
+    double a2 = 0.5 * alpha;
+    
+    for (unsigned i = 0; i < n; ++i) {
+        double x = static_cast<double>(i) / static_cast<double>(n);
+        double window = a0 - a1 * cos(2.0 * M_PI * x) + a2 * cos(4.0 * M_PI * x);
+        p[i] *= float(window);
+    }
+}
+
+} // namespace
+
+void RealtimeAnalyser::doFFTAnalysis()
+{    
+    ASSERT(isMainThread());
+
+    // Unroll the input buffer into a temporary buffer, where we'll apply an analysis window followed by an FFT.
+    size_t fftSize = this->fftSize();
+    
+    AudioFloatArray temporaryBuffer(fftSize);
+    float* inputBuffer = m_inputBuffer.data();
+    float* tempP = temporaryBuffer.data();
+
+    // Take the previous fftSize values from the input buffer and copy into the temporary buffer.
+    // FIXME : optimize with memcpy().
+    unsigned writeIndex = m_writeIndex;
+    for (unsigned i = 0; i < fftSize; ++i)
+        tempP[i] = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
+    
+    // Window the input samples.
+    applyWindow(tempP, fftSize);
+    
+    // Do the analysis.
+    m_analysisFrame->doFFT(tempP);
+
+    size_t n = DefaultFFTSize / 2;
+
+    float* realP = m_analysisFrame->realData();
+    float* imagP = m_analysisFrame->imagData();
+
+    // Blow away the packed nyquist component.
+    imagP[0] = 0.0f;
+    
+    // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
+    const double MagnitudeScale = 1.0 / DefaultFFTSize;
+
+    // A value of 0 does no averaging with the previous result.  Larger values produce slower, but smoother changes.
+    double k = m_smoothingTimeConstant;
+    k = max(0.0, k);
+    k = min(1.0, k);    
+    
+    // Convert the analysis data from complex to magnitude and average with the previous result.
+    float* destination = magnitudeBuffer().data();
+    for (unsigned i = 0; i < n; ++i) {
+        Complex c(realP[i], imagP[i]);
+        double scalarMagnitude = abs(c) * MagnitudeScale;        
+        destination[i] = float(k * destination[i] + (1.0 - k) * scalarMagnitude);
+    }
+}
+
+#if ENABLE(3D_CANVAS)
+
+void RealtimeAnalyser::getFloatFrequencyData(Float32Array* destinationArray)
+{
+    ASSERT(isMainThread());
+
+    if (!destinationArray)
+        return;
+        
+    doFFTAnalysis();
+    
+    // Convert from linear magnitude to floating-point decibels.
+    const double MinDecibels = m_minDecibels;
+    unsigned sourceLength = magnitudeBuffer().size();
+    size_t len = min(sourceLength, destinationArray->length());
+    if (len > 0) {
+        const float* source = magnitudeBuffer().data();
+        float* destination = destinationArray->data();
+        
+        for (unsigned i = 0; i < len; ++i) {
+            float linearValue = source[i];
+            double dbMag = !linearValue ? MinDecibels : AudioUtilities::linearToDecibels(linearValue);
+            destination[i] = float(dbMag);
+        }
+    }
+}
+
+void RealtimeAnalyser::getByteFrequencyData(Uint8Array* destinationArray)
+{
+    ASSERT(isMainThread());
+
+    if (!destinationArray)
+        return;
+        
+    doFFTAnalysis();
+    
+    // Convert from linear magnitude to unsigned-byte decibels.
+    unsigned sourceLength = magnitudeBuffer().size();
+    size_t len = min(sourceLength, destinationArray->length());
+    if (len > 0) {
+        const double RangeScaleFactor = m_maxDecibels == m_minDecibels ? 1.0 : 1.0 / (m_maxDecibels - m_minDecibels);
+
+        const float* source = magnitudeBuffer().data();
+        unsigned char* destination = destinationArray->data();        
+        
+        for (unsigned i = 0; i < len; ++i) {
+            float linearValue = source[i];
+            double dbMag = !linearValue ? m_minDecibels : AudioUtilities::linearToDecibels(linearValue);
+            
+            // The range m_minDecibels to m_maxDecibels will be scaled to byte values from 0 to UCHAR_MAX.
+            double scaledValue = UCHAR_MAX * (dbMag - m_minDecibels) * RangeScaleFactor;
+
+            // Clip to valid range.
+            if (scaledValue < 0.0)
+                scaledValue = 0.0;
+            if (scaledValue > UCHAR_MAX)
+                scaledValue = UCHAR_MAX;
+            
+            destination[i] = static_cast<unsigned char>(scaledValue);
+        }
+    }
+}
+
+void RealtimeAnalyser::getByteTimeDomainData(Uint8Array* destinationArray)
+{
+    ASSERT(isMainThread());
+
+    if (!destinationArray)
+        return;
+        
+    unsigned fftSize = this->fftSize();
+    size_t len = min(fftSize, destinationArray->length());
+    if (len > 0) {
+        bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize && m_inputBuffer.size() > fftSize;
+        ASSERT(isInputBufferGood);
+        if (!isInputBufferGood)
+            return;
+
+        float* inputBuffer = m_inputBuffer.data();        
+        unsigned char* destination = destinationArray->data();
+        
+        unsigned writeIndex = m_writeIndex;
+
+        for (unsigned i = 0; i < len; ++i) {
+            // Buffer access is protected due to modulo operation.
+            float value = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
+
+            // Scale from nominal -1.0 -> +1.0 to unsigned byte.
+            double scaledValue = 128.0 * (value + 1.0);
+
+            // Clip to valid range.
+            if (scaledValue < 0.0)
+                scaledValue = 0.0;
+            if (scaledValue > UCHAR_MAX)
+                scaledValue = UCHAR_MAX;
+            
+            destination[i] = static_cast<unsigned char>(scaledValue);
+        }
+    }
+}
+
+#endif // 3D_CANVAS
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_AUDIO)