/*
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#include "config.h"

#if ENABLE(WEB_AUDIO)

#include "platform/audio/ReverbConvolver.h"

#include "platform/Task.h"
#include "platform/audio/AudioBus.h"
#include "platform/audio/VectorMath.h"
#include "public/platform/Platform.h"
#include "public/platform/WebThread.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 = 128;
const size_t MaxRealtimeFFTSize = 2048;

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_minFFTSize(MinFFTSize) // First stage will have this size - successive stages will double in size each time
    , m_maxFFTSize(maxFFTSize) // until we hit m_maxFFTSize
{
    // 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;

    const float* response = impulseResponse->data();
    size_t totalResponseLength = impulseResponse->length();

    // The total latency is zero because the direct-convolution is used in the leading portion.
    size_t reverbTotalLatency = 0;

    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;

        bool useDirectConvolver = !stageOffset;

        OwnPtr<ReverbConvolverStage> stage = adoptPtr(new ReverbConvolverStage(response, totalResponseLength, reverbTotalLatency, stageOffset, stageSize, fftSize, renderPhase, renderSliceSize, &m_accumulationBuffer, useDirectConvolver));

        bool isBackgroundStage = false;

        if (useBackgroundThreads && stageOffset > RealtimeFrameLimit) {
            m_backgroundStages.append(stage.release());
            isBackgroundStage = true;
        } else
            m_stages.append(stage.release());

        stageOffset += stageSize;
        ++i;

        if (!useDirectConvolver) {
            // 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 (useBackgroundThreads && m_backgroundStages.size() > 0)
        m_backgroundThread = adoptPtr(blink::Platform::current()->createThread("Reverb convolution background thread"));
}

ReverbConvolver::~ReverbConvolver()
{
    // Wait for background thread to stop
    m_backgroundThread.clear();
}

void ReverbConvolver::processInBackground()
{
    // 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(const AudioChannel* sourceChannel, AudioChannel* destinationChannel, size_t framesToProcess)
{
    bool isSafe = sourceChannel && destinationChannel && sourceChannel->length() >= framesToProcess && destinationChannel->length() >= framesToProcess;
    ASSERT(isSafe);
    if (!isSafe)
        return;

    const float* source = sourceChannel->data();
    float* destination = destinationChannel->mutableData();
    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, post another task to the background thread.
    if (m_backgroundThread)
        m_backgroundThread->postTask(new Task(WTF::bind(&ReverbConvolver::processInBackground, this)));
}

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();
}

size_t ReverbConvolver::latencyFrames() const
{
    return 0;
}

} // namespace WebCore

#endif // ENABLE(WEB_AUDIO)