JUCE/modules/juce_audio_devices/sources/juce_AudioTransportSource.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.

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

AudioTransportSource::AudioTransportSource()
    : source (nullptr),
      resamplerSource (nullptr),
      bufferingSource (nullptr),
      positionableSource (nullptr),
      masterSource (nullptr),
      gain (1.0f),
      lastGain (1.0f),
      playing (false),
      stopped (true),
      sampleRate (44100.0),
      sourceSampleRate (0.0),
      blockSize (128),
      readAheadBufferSize (0),
      isPrepared (false),
      inputStreamEOF (false)
{
}

AudioTransportSource::~AudioTransportSource()
{
    setSource (nullptr);
    releaseMasterResources();
}

void AudioTransportSource::setSource (PositionableAudioSource* const newSource,
                                      int readAheadSize, TimeSliceThread* readAheadThread,
                                      double sourceSampleRateToCorrectFor, int maxNumChannels)
{
    if (source == newSource)
    {
        if (source == nullptr)
            return;

        setSource (nullptr, 0, nullptr); // deselect and reselect to avoid releasing resources wrongly
    }

    readAheadBufferSize = readAheadSize;
    sourceSampleRate = sourceSampleRateToCorrectFor;

    ResamplingAudioSource* newResamplerSource = nullptr;
    BufferingAudioSource* newBufferingSource = nullptr;
    PositionableAudioSource* newPositionableSource = nullptr;
    AudioSource* newMasterSource = nullptr;

    ScopedPointer<ResamplingAudioSource> oldResamplerSource (resamplerSource);
    ScopedPointer<BufferingAudioSource> oldBufferingSource (bufferingSource);
    AudioSource* oldMasterSource = masterSource;

    if (newSource != nullptr)
    {
        newPositionableSource = newSource;

        if (readAheadSize > 0)
        {
            // If you want to use a read-ahead buffer, you must also provide a TimeSliceThread
            // for it to use!
            jassert (readAheadThread != nullptr);

            newPositionableSource = newBufferingSource
                = new BufferingAudioSource (newPositionableSource, *readAheadThread,
                                            false, readAheadSize, maxNumChannels);
        }

        newPositionableSource->setNextReadPosition (0);

        if (sourceSampleRateToCorrectFor > 0)
            newMasterSource = newResamplerSource
                = new ResamplingAudioSource (newPositionableSource, false, maxNumChannels);
        else
            newMasterSource = newPositionableSource;

        if (isPrepared)
        {
            if (newResamplerSource != nullptr && sourceSampleRate > 0 && sampleRate > 0)
                newResamplerSource->setResamplingRatio (sourceSampleRate / sampleRate);

            newMasterSource->prepareToPlay (blockSize, sampleRate);
        }
    }

    {
        const ScopedLock sl (callbackLock);

        source = newSource;
        resamplerSource = newResamplerSource;
        bufferingSource = newBufferingSource;
        masterSource = newMasterSource;
        positionableSource = newPositionableSource;

        inputStreamEOF = false;
        playing = false;
    }

    if (oldMasterSource != nullptr)
        oldMasterSource->releaseResources();
}

void AudioTransportSource::start()
{
    if ((! playing) && masterSource != nullptr)
    {
        {
            const ScopedLock sl (callbackLock);
            playing = true;
            stopped = false;
            inputStreamEOF = false;
        }

        sendChangeMessage();
    }
}

void AudioTransportSource::stop()
{
    if (playing)
    {
        {
            const ScopedLock sl (callbackLock);
            playing = false;
        }

        int n = 500;
        while (--n >= 0 && ! stopped)
            Thread::sleep (2);

        sendChangeMessage();
    }
}

void AudioTransportSource::setPosition (double newPosition)
{
    if (sampleRate > 0.0)
        setNextReadPosition ((int64) (newPosition * sampleRate));
}

double AudioTransportSource::getCurrentPosition() const
{
    if (sampleRate > 0.0)
        return (double) getNextReadPosition() / sampleRate;

    return 0.0;
}

double AudioTransportSource::getLengthInSeconds() const
{
    if (sampleRate > 0.0)
        return (double) getTotalLength() / sampleRate;

    return 0.0;
}

void AudioTransportSource::setNextReadPosition (int64 newPosition)
{
    if (positionableSource != nullptr)
    {
        if (sampleRate > 0 && sourceSampleRate > 0)
            newPosition = (int64) ((double) newPosition * sourceSampleRate / sampleRate);

        positionableSource->setNextReadPosition (newPosition);

        if (resamplerSource != nullptr)
            resamplerSource->flushBuffers();

        inputStreamEOF = false;
    }
}

int64 AudioTransportSource::getNextReadPosition() const
{
    if (positionableSource != nullptr)
    {
        const double ratio = (sampleRate > 0 && sourceSampleRate > 0) ? sampleRate / sourceSampleRate : 1.0;
        return (int64) ((double) positionableSource->getNextReadPosition() * ratio);
    }

    return 0;
}

int64 AudioTransportSource::getTotalLength() const
{
    const ScopedLock sl (callbackLock);

    if (positionableSource != nullptr)
    {
        const double ratio = (sampleRate > 0 && sourceSampleRate > 0) ? sampleRate / sourceSampleRate : 1.0;
        return (int64) ((double) positionableSource->getTotalLength() * ratio);
    }

    return 0;
}

bool AudioTransportSource::isLooping() const
{
    const ScopedLock sl (callbackLock);
    return positionableSource != nullptr && positionableSource->isLooping();
}

void AudioTransportSource::setGain (const float newGain) noexcept
{
    gain = newGain;
}

void AudioTransportSource::prepareToPlay (int samplesPerBlockExpected, double newSampleRate)
{
    const ScopedLock sl (callbackLock);

    sampleRate = newSampleRate;
    blockSize = samplesPerBlockExpected;

    if (masterSource != nullptr)
        masterSource->prepareToPlay (samplesPerBlockExpected, sampleRate);

    if (resamplerSource != nullptr && sourceSampleRate > 0)
        resamplerSource->setResamplingRatio (sourceSampleRate / sampleRate);

    inputStreamEOF = false;
    isPrepared = true;
}

void AudioTransportSource::releaseMasterResources()
{
    const ScopedLock sl (callbackLock);

    if (masterSource != nullptr)
        masterSource->releaseResources();

    isPrepared = false;
}

void AudioTransportSource::releaseResources()
{
    releaseMasterResources();
}

void AudioTransportSource::getNextAudioBlock (const AudioSourceChannelInfo& info)
{
    const ScopedLock sl (callbackLock);

    if (masterSource != nullptr && ! stopped)
    {
        masterSource->getNextAudioBlock (info);

        if (! playing)
        {
            // just stopped playing, so fade out the last block..
            for (int i = info.buffer->getNumChannels(); --i >= 0;)
                info.buffer->applyGainRamp (i, info.startSample, jmin (256, info.numSamples), 1.0f, 0.0f);

            if (info.numSamples > 256)
                info.buffer->clear (info.startSample + 256, info.numSamples - 256);
        }

        if (positionableSource->getNextReadPosition() > positionableSource->getTotalLength() + 1
              && ! positionableSource->isLooping())
        {
            playing = false;
            inputStreamEOF = true;
            sendChangeMessage();
        }

        stopped = ! playing;

        for (int i = info.buffer->getNumChannels(); --i >= 0;)
            info.buffer->applyGainRamp (i, info.startSample, info.numSamples, lastGain, gain);
    }
    else
    {
        info.clearActiveBufferRegion();
        stopped = true;
    }

    lastGain = gain;
}