JUCE/modules/juce_audio_devices/sources/juce_AudioSourcePlayer.cpp

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

   This file is part of the JUCE library - "Jules' Utility Class Extensions"
   Copyright 2004-11 by Raw Material Software Ltd.

  ------------------------------------------------------------------------------

   JUCE can be redistributed and/or modified under the terms of the GNU General
   Public License (Version 2), as published by the Free Software Foundation.
   A copy of the license is included in the JUCE distribution, or can be found
   online at www.gnu.org/licenses.

   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.

  ------------------------------------------------------------------------------

   To release a closed-source product which uses JUCE, commercial licenses are
   available: visit www.rawmaterialsoftware.com/juce for more information.

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

AudioSourcePlayer::AudioSourcePlayer()
    : source (nullptr),
      sampleRate (0),
      bufferSize (0),
      tempBuffer (2, 8),
      lastGain (1.0f),
      gain (1.0f)
{
}

AudioSourcePlayer::~AudioSourcePlayer()
{
    setSource (nullptr);
}

void AudioSourcePlayer::setSource (AudioSource* newSource)
{
    if (source != newSource)
    {
        AudioSource* const oldSource = source;

        if (newSource != nullptr && bufferSize > 0 && sampleRate > 0)
            newSource->prepareToPlay (bufferSize, sampleRate);

        {
            const ScopedLock sl (readLock);
            source = newSource;
        }

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

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

void AudioSourcePlayer::audioDeviceIOCallback (const float** inputChannelData,
                                               int totalNumInputChannels,
                                               float** outputChannelData,
                                               int totalNumOutputChannels,
                                               int numSamples)
{
    // these should have been prepared by audioDeviceAboutToStart()...
    jassert (sampleRate > 0 && bufferSize > 0);

    const ScopedLock sl (readLock);

    if (source != nullptr)
    {
        int i, numActiveChans = 0, numInputs = 0, numOutputs = 0;

        // messy stuff needed to compact the channels down into an array
        // of non-zero pointers..
        for (i = 0; i < totalNumInputChannels; ++i)
        {
            if (inputChannelData[i] != nullptr)
            {
                inputChans [numInputs++] = inputChannelData[i];
                if (numInputs >= numElementsInArray (inputChans))
                    break;
            }
        }

        for (i = 0; i < totalNumOutputChannels; ++i)
        {
            if (outputChannelData[i] != nullptr)
            {
                outputChans [numOutputs++] = outputChannelData[i];
                if (numOutputs >= numElementsInArray (outputChans))
                    break;
            }
        }

        if (numInputs > numOutputs)
        {
            // if there aren't enough output channels for the number of
            // inputs, we need to create some temporary extra ones (can't
            // use the input data in case it gets written to)
            tempBuffer.setSize (numInputs - numOutputs, numSamples,
                                false, false, true);

            for (i = 0; i < numOutputs; ++i)
            {
                channels[numActiveChans] = outputChans[i];
                memcpy (channels[numActiveChans], inputChans[i], sizeof (float) * (size_t) numSamples);
                ++numActiveChans;
            }

            for (i = numOutputs; i < numInputs; ++i)
            {
                channels[numActiveChans] = tempBuffer.getSampleData (i - numOutputs, 0);
                memcpy (channels[numActiveChans], inputChans[i], sizeof (float) * (size_t) numSamples);
                ++numActiveChans;
            }
        }
        else
        {
            for (i = 0; i < numInputs; ++i)
            {
                channels[numActiveChans] = outputChans[i];
                memcpy (channels[numActiveChans], inputChans[i], sizeof (float) * (size_t) numSamples);
                ++numActiveChans;
            }

            for (i = numInputs; i < numOutputs; ++i)
            {
                channels[numActiveChans] = outputChans[i];
                zeromem (channels[numActiveChans], sizeof (float) * (size_t) numSamples);
                ++numActiveChans;
            }
        }

        AudioSampleBuffer buffer (channels, numActiveChans, numSamples);

        AudioSourceChannelInfo info (&buffer, 0, numSamples);
        source->getNextAudioBlock (info);

        for (i = info.buffer->getNumChannels(); --i >= 0;)
            buffer.applyGainRamp (i, info.startSample, info.numSamples, lastGain, gain);

        lastGain = gain;
    }
    else
    {
        for (int i = 0; i < totalNumOutputChannels; ++i)
            if (outputChannelData[i] != nullptr)
                zeromem (outputChannelData[i], sizeof (float) * (size_t) numSamples);
    }
}

void AudioSourcePlayer::audioDeviceAboutToStart (AudioIODevice* device)
{
    sampleRate = device->getCurrentSampleRate();
    bufferSize = device->getCurrentBufferSizeSamples();
    zeromem (channels, sizeof (channels));

    if (source != nullptr)
        source->prepareToPlay (bufferSize, sampleRate);
}

void AudioSourcePlayer::audioDeviceStopped()
{
    if (source != nullptr)
        source->releaseResources();

    sampleRate = 0.0;
    bufferSize = 0;

    tempBuffer.setSize (2, 8);
}