Merge WebKit at r75315: Initial merge by git.

Change-Id: I570314b346ce101c935ed22a626b48c2af266b84

1 files changed, 230 insertions, 0 deletions

diff --git a/Source/WebCore/platform/audio/ReverbConvolver.cpp b/Source/WebCore/platform/audio/ReverbConvolver.cpp
new file mode 100644
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+++ b/Source/WebCore/platform/audio/ReverbConvolver.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.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") 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 APPLE 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 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 "ReverbConvolver.h"
+
+#include "VectorMath.h"
+#include "AudioBus.h"
+
+namespace WebCore {
+
+using namespace VectorMath;
+
+const int InputBufferSize = 8 * 16384;
+
+// We only process the leading portion of the impulse response in the real-time thread.  We don't exceed this length.
+// It turns out then, that the background thread has about 278msec of scheduling slop.
+// Empirically, this has been found to be a good compromise between giving enough time for scheduling slop,
+// while still minimizing the amount of processing done in the primary (high-priority) thread.
+// This was found to be a good value on Mac OS X, and may work well on other platforms as well, assuming
+// the very rough scheduling latencies are similar on these time-scales.  Of course, this code may need to be
+// tuned for individual platforms if this assumption is found to be incorrect.
+const size_t RealtimeFrameLimit = 8192  + 4096; // ~278msec @ 44.1KHz
+
+const size_t MinFFTSize = 256;
+const size_t MaxRealtimeFFTSize = 2048;
+
+static void* backgroundThreadEntry(void* threadData)
+{
+    ReverbConvolver* reverbConvolver = static_cast<ReverbConvolver*>(threadData);
+    reverbConvolver->backgroundThreadEntry();
+    return 0;
+}
+
+ReverbConvolver::ReverbConvolver(AudioChannel* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t convolverRenderPhase, bool useBackgroundThreads)
+    : m_impulseResponseLength(impulseResponse->length())
+    , m_accumulationBuffer(impulseResponse->length() + renderSliceSize)
+    , m_inputBuffer(InputBufferSize)
+    , m_renderSliceSize(renderSliceSize)
+    , m_minFFTSize(MinFFTSize) // First stage will have this size - successive stages will double in size each time
+    , m_maxFFTSize(maxFFTSize) // until we hit m_maxFFTSize
+    , m_useBackgroundThreads(useBackgroundThreads)
+    , m_backgroundThread(0)
+    , m_wantsToExit(false)
+    , m_moreInputBuffered(false)
+{
+    // If we are using background threads then don't exceed this FFT size for the
+    // stages which run in the real-time thread.  This avoids having only one or two
+    // large stages (size 16384 or so) at the end which take a lot of time every several
+    // processing slices.  This way we amortize the cost over more processing slices.
+    m_maxRealtimeFFTSize = MaxRealtimeFFTSize;
+
+    // For the moment, a good way to know if we have real-time constraint is to check if we're using background threads.
+    // Otherwise, assume we're being run from a command-line tool.
+    bool hasRealtimeConstraint = useBackgroundThreads;
+
+    float* response = impulseResponse->data();
+    size_t totalResponseLength = impulseResponse->length();
+
+    // Because we're not using direct-convolution in the leading portion, the reverb has an overall latency of half the first-stage FFT size
+    size_t reverbTotalLatency = m_minFFTSize / 2;
+
+    size_t stageOffset = 0;
+    int i = 0;
+    size_t fftSize = m_minFFTSize;
+    while (stageOffset < totalResponseLength) {
+        size_t stageSize = fftSize / 2;
+
+        // For the last stage, it's possible that stageOffset is such that we're straddling the end
+        // of the impulse response buffer (if we use stageSize), so reduce the last stage's length...
+        if (stageSize + stageOffset > totalResponseLength)
+            stageSize = totalResponseLength - stageOffset;
+
+        // This "staggers" the time when each FFT happens so they don't all happen at the same time
+        int renderPhase = convolverRenderPhase + i * renderSliceSize;
+
+        OwnPtr<ReverbConvolverStage> stage(new ReverbConvolverStage(response, totalResponseLength, reverbTotalLatency, stageOffset, stageSize, fftSize, renderPhase, renderSliceSize, &m_accumulationBuffer));
+
+        bool isBackgroundStage = false;
+
+        if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) {
+            m_backgroundStages.append(stage.release());
+            isBackgroundStage = true;
+        } else
+            m_stages.append(stage.release());
+
+        stageOffset += stageSize;
+        ++i;
+
+        // Figure out next FFT size
+        fftSize *= 2;
+        if (hasRealtimeConstraint && !isBackgroundStage && fftSize > m_maxRealtimeFFTSize)
+            fftSize = m_maxRealtimeFFTSize;
+        if (fftSize > m_maxFFTSize)
+            fftSize = m_maxFFTSize;
+    }
+
+    // Start up background thread
+    // FIXME: would be better to up the thread priority here.  It doesn't need to be real-time, but higher than the default...
+    if (this->useBackgroundThreads() && m_backgroundStages.size() > 0)
+        m_backgroundThread = createThread(WebCore::backgroundThreadEntry, this, "convolution background thread");
+}
+
+ReverbConvolver::~ReverbConvolver()
+{
+    // Wait for background thread to stop
+    if (useBackgroundThreads() && m_backgroundThread) {
+        m_wantsToExit = true;
+
+        // Wake up thread so it can return
+        {
+            MutexLocker locker(m_backgroundThreadLock);
+            m_moreInputBuffered = true;
+            m_backgroundThreadCondition.signal();
+        }
+
+        waitForThreadCompletion(m_backgroundThread, 0);
+    }
+}
+
+void ReverbConvolver::backgroundThreadEntry()
+{
+    while (!m_wantsToExit) {
+        // Wait for realtime thread to give us more input
+        m_moreInputBuffered = false;        
+        {
+            MutexLocker locker(m_backgroundThreadLock);
+            while (!m_moreInputBuffered && !m_wantsToExit)
+                m_backgroundThreadCondition.wait(m_backgroundThreadLock);
+        }
+
+        // Process all of the stages until their read indices reach the input buffer's write index
+        int writeIndex = m_inputBuffer.writeIndex();
+
+        // Even though it doesn't seem like every stage needs to maintain its own version of readIndex 
+        // we do this in case we want to run in more than one background thread.
+        int readIndex;
+
+        while ((readIndex = m_backgroundStages[0]->inputReadIndex()) != writeIndex) { // FIXME: do better to detect buffer overrun...
+            // The ReverbConvolverStages need to process in amounts which evenly divide half the FFT size
+            const int SliceSize = MinFFTSize / 2;
+
+            // Accumulate contributions from each stage
+            for (size_t i = 0; i < m_backgroundStages.size(); ++i)
+                m_backgroundStages[i]->processInBackground(this, SliceSize);
+        }
+    }
+}
+
+void ReverbConvolver::process(AudioChannel* sourceChannel, AudioChannel* destinationChannel, size_t framesToProcess)
+{
+    bool isSafe = sourceChannel && destinationChannel && sourceChannel->length() >= framesToProcess && destinationChannel->length() >= framesToProcess;
+    ASSERT(isSafe);
+    if (!isSafe)
+        return;
+        
+    float* source = sourceChannel->data();
+    float* destination = destinationChannel->data();
+    bool isDataSafe = source && destination;
+    ASSERT(isDataSafe);
+    if (!isDataSafe)
+        return;
+
+    // Feed input buffer (read by all threads)
+    m_inputBuffer.write(source, framesToProcess);
+
+    // Accumulate contributions from each stage
+    for (size_t i = 0; i < m_stages.size(); ++i)
+        m_stages[i]->process(source, framesToProcess);
+
+    // Finally read from accumulation buffer
+    m_accumulationBuffer.readAndClear(destination, framesToProcess);
+        
+    // Now that we've buffered more input, wake up our background thread.
+    
+    // Not using a MutexLocker looks strange, but we use a tryLock() instead because this is run on the real-time
+    // thread where it is a disaster for the lock to be contended (causes audio glitching).  It's OK if we fail to
+    // signal from time to time, since we'll get to it the next time we're called.  We're called repeatedly
+    // and frequently (around every 3ms).  The background thread is processing well into the future and has a considerable amount of 
+    // leeway here...
+    if (m_backgroundThreadLock.tryLock()) {
+        m_moreInputBuffered = true;
+        m_backgroundThreadCondition.signal();
+        m_backgroundThreadLock.unlock();
+    }
+}
+
+void ReverbConvolver::reset()
+{
+    for (size_t i = 0; i < m_stages.size(); ++i)
+        m_stages[i]->reset();
+
+    for (size_t i = 0; i < m_backgroundStages.size(); ++i)
+        m_backgroundStages[i]->reset();
+
+    m_accumulationBuffer.reset();
+    m_inputBuffer.reset();
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_AUDIO)