JUCE/modules/juce_audio_basics/sources/juce_BufferingAudioSource.cpp

/*
  ==============================================================================

   This file is part of the JUCE library.
   Copyright (c) 2017 - ROLI Ltd.

   JUCE is an open source library subject to commercial or open-source
   licensing.

   The code included in this file is provided under the terms of the ISC license
   http://www.isc.org/downloads/software-support-policy/isc-license. Permission
   To use, copy, modify, and/or distribute this software for any purpose with or
   without fee is hereby granted provided that the above copyright notice and
   this permission notice appear in all copies.

   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
   DISCLAIMED.

  ==============================================================================
*/

BufferingAudioSource::BufferingAudioSource (PositionableAudioSource* s,
                                            TimeSliceThread& thread,
                                            const bool deleteSourceWhenDeleted,
                                            const int bufferSizeSamples,
                                            const int numChannels,
                                            bool prefillBufferOnPrepareToPlay)
    : source (s, deleteSourceWhenDeleted),
      backgroundThread (thread),
      numberOfSamplesToBuffer (jmax (1024, bufferSizeSamples)),
      numberOfChannels (numChannels),
      bufferValidStart (0),
      bufferValidEnd (0),
      nextPlayPos (0),
      sampleRate (0),
      wasSourceLooping (false),
      isPrepared (false),
      prefillBuffer (prefillBufferOnPrepareToPlay)
{
    jassert (source != nullptr);

    jassert (numberOfSamplesToBuffer > 1024); // not much point using this class if you're
                                              //  not using a larger buffer..
}

BufferingAudioSource::~BufferingAudioSource()
{
    releaseResources();
}

//==============================================================================
void BufferingAudioSource::prepareToPlay (int samplesPerBlockExpected, double newSampleRate)
{
    const int bufferSizeNeeded = jmax (samplesPerBlockExpected * 2, numberOfSamplesToBuffer);

    if (newSampleRate != sampleRate
         || bufferSizeNeeded != buffer.getNumSamples()
         || ! isPrepared)
    {
        backgroundThread.removeTimeSliceClient (this);

        isPrepared = true;
        sampleRate = newSampleRate;

        source->prepareToPlay (samplesPerBlockExpected, newSampleRate);

        buffer.setSize (numberOfChannels, bufferSizeNeeded);
        buffer.clear();

        bufferValidStart = 0;
        bufferValidEnd = 0;

        backgroundThread.addTimeSliceClient (this);

        do
        {
            backgroundThread.moveToFrontOfQueue (this);
            Thread::sleep (5);
        }
        while (prefillBuffer
         && (bufferValidEnd - bufferValidStart < jmin (((int) newSampleRate) / 4, buffer.getNumSamples() / 2)));
    }
}

void BufferingAudioSource::releaseResources()
{
    isPrepared = false;
    backgroundThread.removeTimeSliceClient (this);

    buffer.setSize (numberOfChannels, 0);

    // MSVC2015 seems to need this if statement to not generate a warning during linking.
    // As source is set in the constructor, there is no way that source could
    // ever equal this, but it seems to make MSVC2015 happy.
    if (source != this)
        source->releaseResources();
}

void BufferingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& info)
{
    const ScopedLock sl (bufferStartPosLock);

    const int validStart = (int) (jlimit (bufferValidStart, bufferValidEnd, nextPlayPos) - nextPlayPos);
    const int validEnd   = (int) (jlimit (bufferValidStart, bufferValidEnd, nextPlayPos + info.numSamples) - nextPlayPos);

    if (validStart == validEnd)
    {
        // total cache miss
        info.clearActiveBufferRegion();
    }
    else
    {
        if (validStart > 0)
            info.buffer->clear (info.startSample, validStart);  // partial cache miss at start

        if (validEnd < info.numSamples)
            info.buffer->clear (info.startSample + validEnd,
                                info.numSamples - validEnd);    // partial cache miss at end

        if (validStart < validEnd)
        {
            for (int chan = jmin (numberOfChannels, info.buffer->getNumChannels()); --chan >= 0;)
            {
                jassert (buffer.getNumSamples() > 0);
                const int startBufferIndex = (int) ((validStart + nextPlayPos) % buffer.getNumSamples());
                const int endBufferIndex   = (int) ((validEnd + nextPlayPos)   % buffer.getNumSamples());

                if (startBufferIndex < endBufferIndex)
                {
                    info.buffer->copyFrom (chan, info.startSample + validStart,
                                           buffer,
                                           chan, startBufferIndex,
                                           validEnd - validStart);
                }
                else
                {
                    const int initialSize = buffer.getNumSamples() - startBufferIndex;

                    info.buffer->copyFrom (chan, info.startSample + validStart,
                                           buffer,
                                           chan, startBufferIndex,
                                           initialSize);

                    info.buffer->copyFrom (chan, info.startSample + validStart + initialSize,
                                           buffer,
                                           chan, 0,
                                           (validEnd - validStart) - initialSize);
                }
            }
        }

        nextPlayPos += info.numSamples;
    }
}

bool BufferingAudioSource::waitForNextAudioBlockReady (const AudioSourceChannelInfo& info, const uint32 timeout)
{
    if (!source || source->getTotalLength() <= 0)
        return false;

    if (nextPlayPos + info.numSamples < 0)
        return true;

    if (! isLooping() && nextPlayPos > getTotalLength())
        return true;

    uint32 now = Time::getMillisecondCounter();
    const uint32 startTime = now;

    uint32 elapsed = (now >= startTime ? now - startTime
                                       : (std::numeric_limits<uint32>::max() - startTime) + now);

    while (elapsed <= timeout)
    {
        {
            const ScopedLock sl (bufferStartPosLock);

            const int validStart = static_cast<int> (jlimit (bufferValidStart, bufferValidEnd, nextPlayPos) - nextPlayPos);
            const int validEnd   = static_cast<int> (jlimit (bufferValidStart, bufferValidEnd, nextPlayPos + info.numSamples) - nextPlayPos);

            if (validStart <= 0 && validStart < validEnd && validEnd >= info.numSamples)
                return true;
        }



        if (elapsed < timeout  && (! bufferReadyEvent.wait (static_cast<int> (timeout - elapsed))))
            return false;

        now = Time::getMillisecondCounter();
        elapsed = (now >= startTime ? now - startTime
                                    : (std::numeric_limits<uint32>::max() - startTime) + now);
    }

    return false;
}

int64 BufferingAudioSource::getNextReadPosition() const
{
    jassert (source->getTotalLength() > 0);
    return (source->isLooping() && nextPlayPos > 0)
                    ? nextPlayPos % source->getTotalLength()
                    : nextPlayPos;
}

void BufferingAudioSource::setNextReadPosition (int64 newPosition)
{
    const ScopedLock sl (bufferStartPosLock);

    nextPlayPos = newPosition;
    backgroundThread.moveToFrontOfQueue (this);
}

bool BufferingAudioSource::readNextBufferChunk()
{
    int64 newBVS, newBVE, sectionToReadStart, sectionToReadEnd;

    {
        const ScopedLock sl (bufferStartPosLock);

        if (wasSourceLooping != isLooping())
        {
            wasSourceLooping = isLooping();
            bufferValidStart = 0;
            bufferValidEnd = 0;
        }

        newBVS = jmax ((int64) 0, nextPlayPos);
        newBVE = newBVS + buffer.getNumSamples() - 4;
        sectionToReadStart = 0;
        sectionToReadEnd = 0;

        const int maxChunkSize = 2048;

        if (newBVS < bufferValidStart || newBVS >= bufferValidEnd)
        {
            newBVE = jmin (newBVE, newBVS + maxChunkSize);

            sectionToReadStart = newBVS;
            sectionToReadEnd = newBVE;

            bufferValidStart = 0;
            bufferValidEnd = 0;
        }
        else if (std::abs ((int) (newBVS - bufferValidStart)) > 512
                  || std::abs ((int) (newBVE - bufferValidEnd)) > 512)
        {
            newBVE = jmin (newBVE, bufferValidEnd + maxChunkSize);

            sectionToReadStart = bufferValidEnd;
            sectionToReadEnd = newBVE;

            bufferValidStart = newBVS;
            bufferValidEnd = jmin (bufferValidEnd, newBVE);
        }
    }

    if (sectionToReadStart == sectionToReadEnd)
        return false;

    jassert (buffer.getNumSamples() > 0);
    const int bufferIndexStart = (int) (sectionToReadStart % buffer.getNumSamples());
    const int bufferIndexEnd   = (int) (sectionToReadEnd   % buffer.getNumSamples());

    if (bufferIndexStart < bufferIndexEnd)
    {
        readBufferSection (sectionToReadStart,
                           (int) (sectionToReadEnd - sectionToReadStart),
                           bufferIndexStart);
    }
    else
    {
        const int initialSize = buffer.getNumSamples() - bufferIndexStart;

        readBufferSection (sectionToReadStart,
                           initialSize,
                           bufferIndexStart);

        readBufferSection (sectionToReadStart + initialSize,
                           (int) (sectionToReadEnd - sectionToReadStart) - initialSize,
                           0);
    }

    {
        const ScopedLock sl2 (bufferStartPosLock);

        bufferValidStart = newBVS;
        bufferValidEnd = newBVE;
    }

    bufferReadyEvent.signal();

    return true;
}

void BufferingAudioSource::readBufferSection (const int64 start, const int length, const int bufferOffset)
{
    if (source->getNextReadPosition() != start)
        source->setNextReadPosition (start);

    AudioSourceChannelInfo info (&buffer, bufferOffset, length);
    source->getNextAudioBlock (info);
}

int BufferingAudioSource::useTimeSlice()
{
    return readNextBufferChunk() ? 1 : 100;
}