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#include "config.h"

#if ENABLE(WEB_AUDIO)

#include "ReverbConvolver.h"

#include "VectorMath.h"
#include "AudioBus.h"
#include <mutex>

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
    , m_useBackgroundThreads(useBackgroundThreads)
{
    // 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;

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

        bool isBackgroundStage = false;

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

        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 (this->useBackgroundThreads() && m_backgroundStages.size() > 0) {
        m_backgroundThread = Thread::create("convolution background thread", [this] {
            backgroundThreadEntry();
        });
    }
}

ReverbConvolver::~ReverbConvolver()
{
    // Wait for background thread to stop
    if (useBackgroundThreads() && m_backgroundThread) {
        m_wantsToExit = true;

        // Wake up thread so it can return
        {
            std::lock_guard<Lock> lock(m_backgroundThreadMutex);
            m_moreInputBuffered = true;
            m_backgroundThreadConditionVariable.notifyOne();
        }

        m_backgroundThread->waitForCompletion();
    }
}

void ReverbConvolver::backgroundThreadEntry()
{
    while (!m_wantsToExit) {
        // Wait for realtime thread to give us more input
        m_moreInputBuffered = false;        
        {
            std::unique_lock<Lock> lock(m_backgroundThreadMutex);

            m_backgroundThreadConditionVariable.wait(lock, [this] { return m_moreInputBuffered || m_wantsToExit; });
        }

        // 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, wake up our background thread.
    
    // We use use std::unique_lock with std::try_lock here 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...
    std::unique_lock<Lock> lock(m_backgroundThreadMutex, std::try_to_lock);
    if (!lock.owns_lock())
        return;

    m_moreInputBuffered = true;
    m_backgroundThreadConditionVariable.notifyOne();
}

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)