Source.cpp

#include <stdio.h>
#include <memory.h>
#include <inttypes.h>
#include <vector>
#include <algorithm>
#include <stdlib.h>
 
// typedefs
typedef uint16_t    uint16;
typedef uint32_t    uint32;
typedef int32_t     int32;

const float c_pi = 3.14159265359f;
 
//this struct is the minimal required header data for a wav file
struct SMinimalWaveFileHeader
{
    //the main chunk
    unsigned char m_chunkID[4];
    uint32        m_chunkSize;
    unsigned char m_format[4];
 
    //sub chunk 1 "fmt "
    unsigned char m_subChunk1ID[4];
    uint32        m_subChunk1Size;
    uint16        m_audioFormat;
    uint16        m_numChannels;
    uint32        m_sampleRate;
    uint32        m_byteRate;
    uint16        m_blockAlign;
    uint16        m_bitsPerSample;
 
    //sub chunk 2 "data"
    unsigned char m_subChunk2ID[4];
    uint32        m_subChunk2Size;
 
    //then comes the data!
};

enum class ECrossFade
{
    None,
    In,
    Out,
};

inline void FloatToPCM(unsigned char *PCM, const float& in, size_t numBytes)
{
    // 8 bit is unsigned
    if (numBytes == 1)
    {
        PCM[0] = unsigned char((in * 0.5f + 0.5f) * 255.0f);
        return;
    }

    // casting to double because floats can't exactly store 0x7fffffff, but doubles can.
    // Details of that: https://blog.demofox.org/2017/11/21/floating-point-precision/
    uint32 data;
    if (in < 0.0f)
        data = uint32(double(in) * double(0x80000000));
    else
        data = uint32(double(in) * double(0x7fffffff));

    switch (numBytes)
    {
        case 4: PCM[3] = ((data >> 24) & 0xFF); PCM[2] = ((data >> 16) & 0xFF); PCM[1] = ((data >> 8) & 0xFF); PCM[0] = (data & 0xFF); break;
        case 3: PCM[2] = ((data >> 24) & 0xFF); PCM[1] = ((data >> 16) & 0xFF); PCM[0] = ((data >> 8) & 0xFF); break;
        case 2: PCM[1] = ((data >> 24) & 0xFF); PCM[0] = ((data >> 16) & 0xFF); break;
    }
}

inline void PCMToFloat(float& out, const unsigned char *PCM, size_t numBytes)
{
    // 8 bit is unsigned
    if (numBytes == 1)
    {
        out = (float(PCM[0]) / float(255.0f)) * 2.0f - 1.0f;
        return;
    }

    uint32 data = 0;
    switch (numBytes)
    {
        case 4: data = (uint32(PCM[3]) << 24) | (uint32(PCM[2]) << 16) | (uint32(PCM[1]) << 8) | uint32(PCM[0]); break;
        case 3: data = (uint32(PCM[2]) << 24) | (uint32(PCM[1]) << 16) | (uint32(PCM[0]) << 8); break;
        case 2: data = (uint32(PCM[1]) << 24) | (uint32(PCM[0]) << 16); break;
    }

    // casting to double because floats can't exactly store 0x7fffffff, but doubles can.
    // Details of that: https://blog.demofox.org/2017/11/21/floating-point-precision/
    if (data & 0x80000000)
        out = float(double(int32(data)) / double(0x80000000));
    else
        out = float(double(data) / double(0x7fffffff));
}

// numBytes can be 1, 2, 3, or 4.
// Coresponding to 8 bit, 16 bit, 24 bit, and 32 bit audio.
bool WriteWaveFile(const char *fileName, std::vector<float>& dataFloat, uint16 numChannels, uint32 sampleRate, uint16 numBytes)
{
    std::vector<unsigned char> data;
    data.resize(dataFloat.size() * numBytes);
    for (size_t i = 0; i < dataFloat.size(); ++i)
        FloatToPCM((unsigned char*)&data[i*numBytes], dataFloat[i], numBytes);

    uint32 dataSize = (uint32)data.size();
    uint16 bitsPerSample = numBytes * 8;
 
    //open the file if we can
    FILE *File = nullptr;
    fopen_s(&File, fileName, "w+b");
    if (!File)
    {
        printf("[-----ERROR-----] Could not open %s for writing.\n", fileName);
        return false;
    }
 
    SMinimalWaveFileHeader waveHeader;
 
    //fill out the main chunk
    memcpy(waveHeader.m_chunkID, "RIFF", 4);
    waveHeader.m_chunkSize = dataSize + 36;
    memcpy(waveHeader.m_format, "WAVE", 4);
 
    //fill out sub chunk 1 "fmt "
    memcpy(waveHeader.m_subChunk1ID, "fmt ", 4);
    waveHeader.m_subChunk1Size = 16;
    waveHeader.m_audioFormat = 1;
    waveHeader.m_numChannels = numChannels;
    waveHeader.m_sampleRate = sampleRate;
    waveHeader.m_byteRate = sampleRate * numChannels * bitsPerSample / 8;
    waveHeader.m_blockAlign = numChannels * bitsPerSample / 8;
    waveHeader.m_bitsPerSample = bitsPerSample;
 
    //fill out sub chunk 2 "data"
    memcpy(waveHeader.m_subChunk2ID, "data", 4);
    waveHeader.m_subChunk2Size = dataSize;
 
    //write the header
    fwrite(&waveHeader, sizeof(SMinimalWaveFileHeader), 1, File);
 
    //write the wave data itself
    fwrite(&data[0], dataSize, 1, File);
 
    //close the file and return success
    fclose(File);
    printf("%s saved.\n", fileName);
    return true;
}

bool ReadFileIntoMemory (const char *fileName, std::vector<unsigned char>& data)
{
    //open the file if we can
    FILE *file = nullptr;
    fopen_s(&file, fileName, "rb");
    if (!file)
    {
        printf("[-----ERROR-----]Could not open %s for reading.\n", fileName);
        return false;
    }

    // get the file size and resize the vector to hold the data
    fseek(file, 0, SEEK_END);
    data.resize(ftell(file));

    // read the file into the vector
    fseek(file, 0, SEEK_SET);
    fread(&data[0], 1, data.size(), file);

    // return success
    fclose(file);
    return true;
}

bool ReadWaveFile(const char *fileName, std::vector<float>& data, uint16& numChannels, uint32& sampleRate, uint16& numBytes)
{
    // read the whole file into memory if we can
    std::vector<unsigned char> fileData;
    if (!ReadFileIntoMemory(fileName, fileData))
        return false;
    size_t fileIndex = 0;

	//make sure the main chunk ID is "RIFF"
    if ((fileData.size() < fileIndex + 4) || memcmp(&fileData[fileIndex], "RIFF", 4))
    {
        printf("[-----ERROR-----]%s is an invalid input file. (1)\n", fileName);
        return false;
    }
    fileIndex += 4;

	//get the main chunk size
	uint32 chunkSize;
    if (fileData.size() < fileIndex + 4)
    {
        printf("[-----ERROR-----]%s is an invalid input file. (2)\n", fileName);
        return false;
    }
    chunkSize = *(uint32*)&fileData[fileIndex];
    fileIndex += 4;

	//make sure the format is "WAVE"
    if ((fileData.size() < fileIndex + 4) || memcmp(&fileData[fileIndex], "WAVE", 4))
    {
        printf("[-----ERROR-----]%s is an invalid input file. (3)\n", fileName);
        return false;
    }
    fileIndex += 4;

	size_t chunkPosFmt = -1;
	size_t chunkPosData = -1;
	while(chunkPosFmt == -1 || chunkPosData == -1)
	{
        // get a chunk id and chunk size if we can
        if (fileData.size() < fileIndex + 8)
        {
            printf("[-----ERROR-----]%s is an invalid input file. (4)\n", fileName);
            return false;
        }

        // get the chunk id if we can
        const unsigned char* chunkID = (unsigned char*)&fileData[fileIndex];
        fileIndex += 4;
        chunkSize = *(uint32*)&fileData[fileIndex];
        fileIndex += 4;

		//if we hit a fmt
		if(!memcmp(chunkID,"fmt ", 4))
		{
			chunkPosFmt = (long)(fileIndex - 8);
		}
		//else if we hit a data
		else if(!memcmp(chunkID,"data", 4))
		{
			chunkPosData = (long)(fileIndex - 8);
		}

		//skip to the next chunk
        fileIndex += chunkSize;
	}

	//we'll use this handy struct to load in 
	SMinimalWaveFileHeader waveData;

	//load the fmt part if we can
    fileIndex = chunkPosFmt;
    if (fileData.size() < fileIndex + 24)
    {
        printf("[-----ERROR-----]%s is an invalid input file. (5)\n", fileName);
        return false;
    }
    memcpy(&waveData.m_subChunk1ID, &fileData[fileIndex], 24);
    fileIndex += 24;

	//load the data part if we can
    fileIndex = chunkPosData;
    if (fileData.size() < fileIndex + 8)
    {
        printf("[-----ERROR-----]%s is an invalid input file. (6)\n", fileName);
        return false;
    }
    memcpy(&waveData.m_subChunk2ID, &fileData[fileIndex], 8);
    fileIndex += 8;

	//verify a couple things about the file data
	if(waveData.m_audioFormat != 1 ||       //only pcm data
	   waveData.m_numChannels < 1 ||        //must have a channel
	   waveData.m_numChannels > 2 ||        //must not have more than 2
	   waveData.m_bitsPerSample > 32 ||     //32 bits per sample max
	   waveData.m_bitsPerSample % 8 != 0 || //must be a multiple of 8 bites
	   waveData.m_blockAlign > 8)           //blocks must be 8 bytes or lower
    {
        printf("[-----ERROR-----]%s is an invalid input file. (7)\n", fileName);
        return false;
    }

	//figure out how many samples and blocks there are total in the source data
    size_t bytesPerSample = waveData.m_blockAlign / waveData.m_numChannels;
    size_t numSourceSamples = waveData.m_subChunk2Size / bytesPerSample;

	//allocate space for the source samples
    data.resize(numSourceSamples);

	//read in the source samples at whatever sample rate / number of channels it might be in
    if (fileData.size() < fileIndex + numSourceSamples * bytesPerSample)
    {
        printf("[-----ERROR-----]%s is an invalid input file. (8)\n", fileName);
        return false;
    }

	for(size_t nIndex = 0; nIndex < numSourceSamples; ++nIndex)
	{	
        PCMToFloat(data[nIndex], &fileData[fileIndex], bytesPerSample);
        fileIndex += bytesPerSample;
	}

	//return our data
    numChannels = waveData.m_numChannels;
    sampleRate = waveData.m_sampleRate;
    numBytes = waveData.m_bitsPerSample / 8;

    printf("%s loaded.\n", fileName);
	return true;
}

// Cubic hermite interpolation. More information available here: https://blog.demofox.org/2015/08/08/cubic-hermite-interpolation/
// t is a value that goes from 0 to 1 to interpolate in a C1 continuous way across uniformly sampled data points.
// when t is 0, this will return B.  When t is 1, this will return C.
static float CubicHermite (float A, float B, float C, float D, float t)
{
    float a = -A/2.0f + (3.0f*B)/2.0f - (3.0f*C)/2.0f + D/2.0f;
    float b = A - (5.0f*B)/2.0f + 2.0f*C - D / 2.0f;
    float c = -A/2.0f + C/2.0f;
    float d = B;
 
    return a*t*t*t + b*t*t + c*t + d;
}

inline float SampleChannelFractional (const std::vector<float>& input, float sampleFloat, uint16 channel, uint16 numChannels)
{
    // change this to #if 0 to use linear interpolation instead, which is faster but lower quality
#if 1

    // This uses cubic hermite interpolation to get values between samples

    size_t sample = size_t(sampleFloat);
    float sampleFraction = sampleFloat - std::floorf(sampleFloat);

    size_t sampleIndexNeg1 = (sample > 0) ? sample - 1 : sample;
    size_t sampleIndex0 = sample;
    size_t sampleIndex1 = sample + 1;
    size_t sampleIndex2 = sample + 2;

    sampleIndexNeg1 = sampleIndexNeg1 * numChannels + channel;
    sampleIndex0 = sampleIndex0 * numChannels + channel;
    sampleIndex1 = sampleIndex1 * numChannels + channel;
    sampleIndex2 = sampleIndex2 * numChannels + channel;

    sampleIndexNeg1 = std::min(sampleIndexNeg1, input.size() - 1);
    sampleIndex0 = std::min(sampleIndex0, input.size() - 1);
    sampleIndex1 = std::min(sampleIndex1, input.size() - 1);
    sampleIndex2 = std::min(sampleIndex2, input.size() - 1);

    return CubicHermite(input[sampleIndexNeg1], input[sampleIndex0], input[sampleIndex1], input[sampleIndex2], sampleFraction);
#else

    // This uses linear interpolation to get values between samples.

    size_t sample = size_t(sampleFloat);
    float sampleFraction = sampleFloat - std::floorf(sampleFloat);

    size_t sample1Index = sample * numChannels + channel;
    sample1Index = std::min(sample1Index, input.size() - 1);
    float value1 = input[sample1Index];

    size_t sample2Index = (sample+1) * numChannels + channel;
    sample2Index = std::min(sample2Index, input.size() - 1);
    float value2 = input[sample1Index];

    return value1 * (1.0f - sampleFraction) + value2 * sampleFraction;
#endif
}

// Resample
void TimeAdjust (const std::vector<float>& input, std::vector<float>& output, uint16 numChannels, float timeMultiplier)
{
    size_t numSrcSamples = input.size() / numChannels;
    size_t numOutSamples = (size_t)(float(numSrcSamples) * timeMultiplier);
    output.resize(numOutSamples * numChannels);

    for (size_t outSample = 0; outSample < numOutSamples; ++outSample)
    {
        float percent = float(outSample) / float(numOutSamples-1);

        float srcSampleFloat = float(numSrcSamples) * percent;

        for (uint16 channel = 0; channel < numChannels; ++channel)
            output[outSample*numChannels + channel] = SampleChannelFractional(input, srcSampleFloat, channel, numChannels);
    }
}

// writes a grain to the output buffer, applying a fade in or fade out at the beginning if it should, as well as a pitch multiplier (playback speed multiplier) for the grain
size_t SplatGrainToOutput(const std::vector<float>& input, std::vector<float>& output, uint16 numChannels, size_t grainStart, size_t grainSize, size_t outputSampleIndex, ECrossFade crossFade, size_t crossFadeSize, float pitchMultiplier, bool isFinalGrain)
{
    // calculate starting indices
    size_t outputIndex = outputSampleIndex * numChannels;

    // write the samples
    size_t numSamplesWritten = 0;
    for (float sample = 0; sample < float(grainSize); sample += pitchMultiplier)
    {
        // break out of the loop if we are out of bounds on the input or output
        if (outputIndex + numChannels > output.size())
            break;

        float inputIndexSamples = float(grainStart) + sample;
        if (size_t(inputIndexSamples) * numChannels + numChannels > input.size())
            break;

        // calculate envelope for this sample
        float envelope = 1.0f;
        if (crossFade != ECrossFade::None)
        {
            if (sample <= float(crossFadeSize))
                envelope = sample / float(crossFadeSize);
            if (crossFade == ECrossFade::Out)
                envelope = 1.0f - envelope;
        }

        // write the enveloped sample
        for (uint16 channel = 0; channel < numChannels; ++channel)
            output[outputIndex + channel] += SampleChannelFractional(input, inputIndexSamples, channel, numChannels) * envelope;

        // move to the next samples
        outputIndex += numChannels;
        ++numSamplesWritten;
    }

    // report an error if ever the cross fade size was bigger than the actual grain size, since this causes popping and would be hard to find the cause of.
    // suppress error on final grain since there can be false positives due to sound ending. That makes false negatives but calling this good enough.
    if (!isFinalGrain && crossFadeSize > numSamplesWritten)
    {
        static bool reportedError = false;
        if (!reportedError)
        {
            printf("[-----ERROR-----] cross fade is longer than a grain size! (error only reported once)\n");
            reportedError = true;
        }
    }

    // return how many samples we wrote
    return numSamplesWritten;
}

void GranularTimePitchAdjust (const std::vector<float>& input, std::vector<float>& output, uint16 numChannels, uint32 sampleRate, float timeMultiplier, float pitchMultiplier, float grainSizeSeconds, float crossFadeSeconds)
{
    // calculate size of output buffer and resize it
    size_t numInputSamples = input.size() / numChannels;
    size_t numOutputSamples = (size_t)(float(numInputSamples) * timeMultiplier);
    output.clear();
    output.resize(numOutputSamples * numChannels, 0.0f);

    // calculate how many grains are in the input data
    size_t grainSizeSamples = size_t(float(sampleRate)*grainSizeSeconds);
    size_t numGrains = numInputSamples / grainSizeSamples;
    if (numInputSamples % grainSizeSamples)
        numGrains++;

    // calculate the cross fade size
    size_t crossFadeSizeSamples = size_t(float(sampleRate)*crossFadeSeconds);

    // Repeat each grain 0 or more times to make the output be the correct size
    size_t outputSampleIndex = 0;
    size_t lastGrainWritten = -1;
    for (size_t grain = 0; grain < numGrains; ++grain)
    {
        // calculate the boundaries of the grain
        size_t inputGrainStart = grain * grainSizeSamples;

        // calculate the end of where this grain should go in the output buffer
        size_t outputSampleWindowEnd = size_t(float(inputGrainStart + grainSizeSamples) * timeMultiplier);

        // Splat out zero or more copies of the grain to get our output to be at least as far as we want it to be.
        // Zero copies happens when we shorten time and need to cut pieces (grains) out of the original sound
        while (outputSampleIndex < outputSampleWindowEnd)
        {
            bool isFinalGrain = (grain == numGrains - 1);

            // if we are writing our first grain, or the last grain we wrote was the previous grain, then we don't need to do a cross fade`
            if ((lastGrainWritten == -1) || (lastGrainWritten == grain - 1))
            {
                outputSampleIndex += SplatGrainToOutput(input, output, numChannels, inputGrainStart, grainSizeSamples, outputSampleIndex, ECrossFade::None, crossFadeSizeSamples, pitchMultiplier, isFinalGrain);
                lastGrainWritten = grain;
                continue;
            }

            // else we need to fade out the old grain and then fade in the new one.
            // NOTE: fading out the old grain means starting to play the grain after the last one and bringing it's volume down to zero.
            SplatGrainToOutput(input, output, numChannels, (lastGrainWritten + 1) * grainSizeSamples, grainSizeSamples, outputSampleIndex, ECrossFade::Out, crossFadeSizeSamples, pitchMultiplier, isFinalGrain);
            outputSampleIndex += SplatGrainToOutput(input, output, numChannels, inputGrainStart, grainSizeSamples, outputSampleIndex, ECrossFade::In, crossFadeSizeSamples, pitchMultiplier, isFinalGrain);
            lastGrainWritten = grain;
        }
    }
}

template <typename LAMBDA>
void GranularTimePitchAdjustDynamic (const std::vector<float>& input, std::vector<float>& output, uint16 numChannels, uint32 sampleRate, float grainSizeSeconds, float crossFadeSeconds, const LAMBDA& settingsCallback)
{
    // calculate how many grains are in the input data
    size_t numInputSamples = input.size() / numChannels;
    size_t grainSizeSamples = size_t(float(sampleRate)*grainSizeSeconds);
    size_t numGrains = numInputSamples / grainSizeSamples;
    if (numInputSamples % grainSizeSamples)
        numGrains++;

    // calculate size of output buffer and resize it
    size_t numOutputSamples = 0;
    for (size_t i = 0; i < numGrains; ++i)
    {
        size_t grainStart = i * grainSizeSamples;
        size_t grainEnd = grainStart + grainSizeSamples;
        grainEnd = std::min(grainEnd, input.size());
        size_t grainSize = grainEnd - grainStart;

        float percent = float(i) / float(numGrains);
        float timeMultiplier = 1.0f;
        float pitchMultiplier = 1.0f;
        settingsCallback(percent, timeMultiplier, pitchMultiplier);

        numOutputSamples += (size_t)(float(grainSize) * timeMultiplier);
    }
    output.clear();
    output.resize(numOutputSamples * numChannels, 0.0f);

    // calculate the cross fade size
    size_t crossFadeSizeSamples = size_t(float(sampleRate)*crossFadeSeconds);

    // Repeat each grain 0 or more times to make the output be the correct size
    size_t outputSampleIndex = 0;
    size_t lastGrainWritten = -1;
    float lastGrainPitchMultiplier = 1.0f;
    size_t outputSampleWindowEnd = 0;
    for (size_t grain = 0; grain < numGrains; ++grain)
    {
        // calculate the boundaries of the grain
        size_t inputGrainStart = grain * grainSizeSamples;

        // calculate the end of where this grain should go in the output buffer
        float percent = float(grain) / float(numGrains);
        float timeMultiplier = 1.0f;
        float pitchMultiplier = 1.0f;
        settingsCallback(percent, timeMultiplier, pitchMultiplier);
        outputSampleWindowEnd += size_t(float(grainSizeSamples) * timeMultiplier);

        // Splat out zero or more copies of the grain to get our output to be at least as far as we want it to be.
        // Zero copies happens when we shorten time and need to cut pieces (grains) out of the original sound
        while (outputSampleIndex < outputSampleWindowEnd)
        {
            bool isFinalGrain = (grain == numGrains - 1);

            // if we are writing our first grain, or the last grain we wrote was the previous grain, then we don't need to do a cross fade`
            if ((lastGrainWritten == -1) || (lastGrainWritten == grain - 1))
            {
                outputSampleIndex += SplatGrainToOutput(input, output, numChannels, inputGrainStart, grainSizeSamples, outputSampleIndex, ECrossFade::None, crossFadeSizeSamples, pitchMultiplier, isFinalGrain);
                lastGrainWritten = grain;
                lastGrainPitchMultiplier = pitchMultiplier;
                continue;
            }

            // else we need to fade out the old grain and then fade in the new one.
            // NOTE: fading out the old grain means starting to play the grain after the last one and bringing it's volume down to zero, using the previous grain's pitch multiplier.
            SplatGrainToOutput(input, output, numChannels, (lastGrainWritten + 1) * grainSizeSamples, grainSizeSamples, outputSampleIndex, ECrossFade::Out, crossFadeSizeSamples, lastGrainPitchMultiplier, isFinalGrain);
            outputSampleIndex += SplatGrainToOutput(input, output, numChannels, inputGrainStart, grainSizeSamples, outputSampleIndex, ECrossFade::In, crossFadeSizeSamples, pitchMultiplier, isFinalGrain);
            lastGrainWritten = grain;
            lastGrainPitchMultiplier = pitchMultiplier;
        }
    }
}

//the entry point of our application
int main(int argc, char **argv)
{
    // load the wave file
    uint16 numChannels;
    uint32 sampleRate;
    uint16 numBytes;
    std::vector<float> source, out, sourceLeft, sourceRight;
    ReadWaveFile("legend1.wav", source, numChannels, sampleRate, numBytes);

    // speed up the audio and increase pitch
    {
        TimeAdjust(source, out, numChannels, 0.7f);
        WriteWaveFile("out_A_FastHigh.wav", out, numChannels, sampleRate, numBytes);

        TimeAdjust(source, out, numChannels, 0.4f);
        WriteWaveFile("out_A_FasterHigher.wav", out, numChannels, sampleRate, numBytes);
    }

    // slow down the audio and decrease pitch
    {
        TimeAdjust(source, out, numChannels, 1.3f);
        WriteWaveFile("out_A_SlowLow.wav", out, numChannels, sampleRate, numBytes);

        TimeAdjust(source, out, numChannels, 2.1f);
        WriteWaveFile("out_A_SlowerLower.wav", out, numChannels, sampleRate, numBytes);
    }

    // speed up audio without affecting pitch
    {
        GranularTimePitchAdjust(source, out, numChannels, sampleRate, 0.7f, 1.0f, 0.02f, 0.002f);
        WriteWaveFile("out_B_Fast.wav", out, numChannels, sampleRate, numBytes);

        GranularTimePitchAdjust(source, out, numChannels, sampleRate, 0.4f, 1.0f, 0.02f, 0.002f);
        WriteWaveFile("out_B_Faster.wav", out, numChannels, sampleRate, numBytes);
    }

    // slow down audio without affecting pitch
    {
        GranularTimePitchAdjust(source, out, numChannels, sampleRate, 1.3f, 1.0f, 0.02f, 0.002f);
        WriteWaveFile("out_B_Slow.wav", out, numChannels, sampleRate, numBytes);

        GranularTimePitchAdjust(source, out, numChannels, sampleRate, 2.1f, 1.0f, 0.02f, 0.002f);
        WriteWaveFile("out_B_Slower.wav", out, numChannels, sampleRate, numBytes);
    }

    // Make pitch higher without affecting length
    {
        // do it in two steps - first as a granular time adjust, and then as a pitch/time adjust
        std::vector<float> out2;
        GranularTimePitchAdjust(source, out2, numChannels, sampleRate, 1.0f / 0.7f, 1.0f, 0.02f, 0.002f);
        TimeAdjust(out2, out, numChannels, 0.7f);
        WriteWaveFile("out_C_HighAlternate.wav", out, numChannels, sampleRate, numBytes);

        // do it in one step by changing grain playback speeds
        GranularTimePitchAdjust(source, out, numChannels, sampleRate, 1.0f, 1.0f / 0.7f, 0.02f, 0.002f);
        WriteWaveFile("out_C_High.wav", out, numChannels, sampleRate, numBytes);

        GranularTimePitchAdjust(source, out, numChannels, sampleRate, 1.0f, 1.0f / 0.4f, 0.02f, 0.002f);
        WriteWaveFile("out_C_Higher.wav", out, numChannels, sampleRate, numBytes);
    }

    // make pitch lower without affecting length
    {
        GranularTimePitchAdjust(source, out, numChannels, sampleRate, 1.0f, 1.0f / 1.3f, 0.02f, 0.002f);
        WriteWaveFile("out_C_Low.wav", out, numChannels, sampleRate, numBytes);

        GranularTimePitchAdjust(source, out, numChannels, sampleRate, 1.0f, 1.0f / 2.1f, 0.02f, 0.002f);
        WriteWaveFile("out_C_Lower.wav", out, numChannels, sampleRate, numBytes);
    }

    // Make pitch lower but speed higher
    {
        GranularTimePitchAdjust(source, out, numChannels, sampleRate, 1.3f, 1.0f / 0.7f, 0.02f, 0.002f);
        WriteWaveFile("out_D_SlowHigh.wav", out, numChannels, sampleRate, numBytes);

        GranularTimePitchAdjust(source, out, numChannels, sampleRate, 0.7f, 1.0f / 1.3f, 0.02f, 0.002f);
        WriteWaveFile("out_D_FastLow.wav", out, numChannels, sampleRate, numBytes);
    }

    // dynamic tests which change time and pitch multipliers over time (for each input grain)
    {
        // adjust pitch on a sine wave
        GranularTimePitchAdjustDynamic(source, out, numChannels, sampleRate, 0.02f, 0.002f,
            [] (float percent, float& timeMultiplier, float& pitchMultiplier)
            {
                // time is 1
                // pitch is 10hz from 0.75 to 1.25
                timeMultiplier = 1.0f;
                pitchMultiplier = 1.0f / ((std::sinf(percent * c_pi * 10.0f) * 0.5f + 0.5f) * 0.5f + 0.75f);
            }
        );
        WriteWaveFile("out_E_Pitch.wav", out, numChannels, sampleRate, numBytes);

        // adjust speed on a sine wave
        GranularTimePitchAdjustDynamic(source, out, numChannels, sampleRate, 0.02f, 0.002f,
            [] (float percent, float& timeMultiplier, float& pitchMultiplier)
            {
                // time is 13hz from 0.5 to 2.5
                // pitch is 1
                timeMultiplier = (std::sinf(percent * c_pi * 13.0f) * 0.5f + 0.5f) * 2.0f + 0.5f;
                pitchMultiplier = 1.0f;
            }
        );
        WriteWaveFile("out_E_Time.wav", out, numChannels, sampleRate, numBytes);

        // adjust time and speed on a sine wave
        GranularTimePitchAdjustDynamic(source, out, numChannels, sampleRate, 0.02f, 0.002f,
            [] (float percent, float& timeMultiplier, float& pitchMultiplier)
            {
                // time is 13hz from 0.5 to 2.5
                // pitch is 10hz from 0.75 to 1.25
                timeMultiplier = (std::sinf(percent * c_pi * 10.0f) * 0.5f + 0.5f) * 2.0f + 0.5f;
                pitchMultiplier = 1.0f / ((std::sinf(percent * c_pi * 10.0f) * 0.5f + 0.5f) * 0.5f + 0.75f);
            }
        );
        WriteWaveFile("out_E_TimePitch.wav", out, numChannels, sampleRate, numBytes);
    }

    system("pause");
}