예제 #1
0
MemoryAudioSource::MemoryAudioSource (AudioBuffer<float>& bufferToUse, bool copyMemory, bool shouldLoop)
    : isLooping (shouldLoop)
{
    if (copyMemory)
        buffer.makeCopyOf (bufferToUse);
    else
        buffer.setDataToReferTo (bufferToUse.getArrayOfWritePointers(),
                                 bufferToUse.getNumChannels(),
                                 bufferToUse.getNumSamples());
}
    void perform (AudioBuffer<FloatType>& buffer, MidiBuffer& midiMessages, AudioPlayHead* audioPlayHead)
    {
        auto numSamples = buffer.getNumSamples();
        auto maxSamples = renderingBuffer.getNumSamples();

        if (numSamples > maxSamples)
        {
            // being asked to render more samples than our buffers have, so slice things up...
            tempMIDI.clear();
            tempMIDI.addEvents (midiMessages, maxSamples, numSamples, -maxSamples);

            {
                AudioBuffer<FloatType> startAudio (buffer.getArrayOfWritePointers(), buffer.getNumChannels(), maxSamples);
                midiMessages.clear (maxSamples, numSamples);
                perform (startAudio, midiMessages, audioPlayHead);
            }

            AudioBuffer<FloatType> endAudio (buffer.getArrayOfWritePointers(), buffer.getNumChannels(), maxSamples, numSamples - maxSamples);
            perform (endAudio, tempMIDI, audioPlayHead);
            return;
        }

        currentAudioInputBuffer = &buffer;
        currentAudioOutputBuffer.setSize (jmax (1, buffer.getNumChannels()), numSamples);
        currentAudioOutputBuffer.clear();
        currentMidiInputBuffer = &midiMessages;
        currentMidiOutputBuffer.clear();

        {
            const Context context { renderingBuffer.getArrayOfWritePointers(), midiBuffers.begin(), audioPlayHead, numSamples };

            for (auto* op : renderOps)
                op->perform (context);
        }

        for (int i = 0; i < buffer.getNumChannels(); ++i)
            buffer.copyFrom (i, 0, currentAudioOutputBuffer, i, 0, numSamples);

        midiMessages.clear();
        midiMessages.addEvents (currentMidiOutputBuffer, 0, buffer.getNumSamples(), 0);
        currentAudioInputBuffer = nullptr;
    }
예제 #3
0
void SynthesiserVoice::renderNextBlock (AudioBuffer<double>& outputBuffer,
                                        int startSample, int numSamples)
{
    AudioBuffer<double> subBuffer (outputBuffer.getArrayOfWritePointers(),
                                   outputBuffer.getNumChannels(),
                                   startSample, numSamples);

    tempBuffer.makeCopyOf (subBuffer, true);
    renderNextBlock (tempBuffer, 0, numSamples);
    subBuffer.makeCopyOf (tempBuffer, true);
}
예제 #4
0
void AudioFormatReader::readMaxLevels (int64 startSampleInFile, int64 numSamples,
                                       Range<float>* const results, const int channelsToRead)
{
    jassert (channelsToRead > 0 && channelsToRead <= (int) numChannels);

    if (numSamples <= 0)
    {
        for (int i = 0; i < channelsToRead; ++i)
            results[i] = Range<float>();

        return;
    }

    auto bufferSize = (int) jmin (numSamples, (int64) 4096);
    AudioBuffer<float> tempSampleBuffer ((int) channelsToRead, bufferSize);

    auto floatBuffer = tempSampleBuffer.getArrayOfWritePointers();
    auto intBuffer = reinterpret_cast<int* const*> (floatBuffer);
    bool isFirstBlock = true;

    while (numSamples > 0)
    {
        auto numToDo = (int) jmin (numSamples, (int64) bufferSize);

        if (! read (intBuffer, channelsToRead, startSampleInFile, numToDo, false))
            break;

        for (int i = 0; i < channelsToRead; ++i)
        {
            Range<float> r;

            if (usesFloatingPointData)
            {
                r = FloatVectorOperations::findMinAndMax (floatBuffer[i], numToDo);
            }
            else
            {
                auto intRange = Range<int>::findMinAndMax (intBuffer[i], numToDo);

                r = Range<float> (intRange.getStart() / (float) std::numeric_limits<int>::max(),
                                  intRange.getEnd()   / (float) std::numeric_limits<int>::max());
            }

            results[i] = isFirstBlock ? r : results[i].getUnionWith (r);
        }

        isFirstBlock = false;
        numSamples -= numToDo;
        startSampleInFile += numToDo;
    }
}
예제 #5
0
    bool readFromInputDevice (AudioBuffer<float>& inputChannelBuffer, const int numSamples)
    {
        jassert (numChannelsRunning <= inputChannelBuffer.getNumChannels());
        float* const* const data = inputChannelBuffer.getArrayOfWritePointers();

        if (isInterleaved)
        {
            scratch.ensureSize ((size_t) ((int) sizeof (float) * numSamples * numChannelsRunning), false);
            scratch.fillWith (0); // (not clearing this data causes warnings in valgrind)

            auto num = snd_pcm_readi (handle, scratch.getData(), (snd_pcm_uframes_t) numSamples);

            if (num < 0)
            {
                if (num == -(EPIPE))
                    overrunCount++;

                if (JUCE_ALSA_FAILED (snd_pcm_recover (handle, (int) num, 1 /* silent */)))
                    return false;
            }


            if (num < numSamples)
                JUCE_ALSA_LOG ("Did not read all samples: num: " << num << ", numSamples: " << numSamples);

            for (int i = 0; i < numChannelsRunning; ++i)
                converter->convertSamples (data[i], 0, scratch.getData(), i, numSamples);
        }
        else
        {
            auto num = snd_pcm_readn (handle, (void**) data, (snd_pcm_uframes_t) numSamples);

            if (num < 0)
            {
                if (num == -(EPIPE))
                    overrunCount++;

                if (JUCE_ALSA_FAILED (snd_pcm_recover (handle, (int) num, 1 /* silent */)))
                    return false;
            }

            if (num < numSamples)
                JUCE_ALSA_LOG ("Did not read all samples: num: " << num << ", numSamples: " << numSamples);

            for (int i = 0; i < numChannelsRunning; ++i)
                converter->convertSamples (data[i], data[i], numSamples);
        }

        return true;
    }
예제 #6
0
    //==============================================================================
    bool writeToOutputDevice (AudioBuffer<float>& outputChannelBuffer, const int numSamples)
    {
        jassert (numChannelsRunning <= outputChannelBuffer.getNumChannels());
        float* const* const data = outputChannelBuffer.getArrayOfWritePointers();
        snd_pcm_sframes_t numDone = 0;

        if (isInterleaved)
        {
            scratch.ensureSize ((size_t) ((int) sizeof (float) * numSamples * numChannelsRunning), false);

            for (int i = 0; i < numChannelsRunning; ++i)
                converter->convertSamples (scratch.getData(), i, data[i], 0, numSamples);

            numDone = snd_pcm_writei (handle, scratch.getData(), (snd_pcm_uframes_t) numSamples);
        }
        else
        {
            for (int i = 0; i < numChannelsRunning; ++i)
                converter->convertSamples (data[i], data[i], numSamples);

            numDone = snd_pcm_writen (handle, (void**) data, (snd_pcm_uframes_t) numSamples);
        }

        if (numDone < 0)
        {
            if (numDone == -(EPIPE))
                underrunCount++;

            if (JUCE_ALSA_FAILED (snd_pcm_recover (handle, (int) numDone, 1 /* silent */)))
                return false;
        }

        if (numDone < numSamples)
            JUCE_ALSA_LOG ("Did not write all samples: numDone: " << numDone << ", numSamples: " << numSamples);

        return true;
    }