diff --git a/story-editor/tools/audio/AudioFile.cpp b/story-editor/tools/audio/AudioFile.cpp
new file mode 100755
index 0000000..2d1dc59
--- /dev/null
+++ b/story-editor/tools/audio/AudioFile.cpp
@@ -0,0 +1,881 @@
+//=======================================================================
+/** @file AudioFile.cpp
+ *  @author Adam Stark
+ *  @copyright Copyright (C) 2017  Adam Stark
+ *
+ * This file is part of the 'AudioFile' library
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+//=======================================================================
+
+#include "AudioFile.h"
+#include <fstream>
+#include <unordered_map>
+#include <iterator>
+#include <cstdint>
+
+//=============================================================
+// Pre-defined 10-byte representations of common sample rates
+std::unordered_map <uint32_t, std::vector<uint8_t>> aiffSampleRateTable = {
+    {8000, {64, 11, 250, 0, 0, 0, 0, 0, 0, 0}},
+    {11025, {64, 12, 172, 68, 0, 0, 0, 0, 0, 0}},
+    {16000, {64, 12, 250, 0, 0, 0, 0, 0, 0, 0}},
+    {22050, {64, 13, 172, 68, 0, 0, 0, 0, 0, 0}},
+    {32000, {64, 13, 250, 0, 0, 0, 0, 0, 0, 0}},
+    {37800, {64, 14, 147, 168, 0, 0, 0, 0, 0, 0}},
+    {44056, {64, 14, 172, 24, 0, 0, 0, 0, 0, 0}},
+    {44100, {64, 14, 172, 68, 0, 0, 0, 0, 0, 0}},
+    {47250, {64, 14, 184, 146, 0, 0, 0, 0, 0, 0}},
+    {48000, {64, 14, 187, 128, 0, 0, 0, 0, 0, 0}},
+    {50000, {64, 14, 195, 80, 0, 0, 0, 0, 0, 0}},
+    {50400, {64, 14, 196, 224, 0, 0, 0, 0, 0, 0}},
+    {88200, {64, 15, 172, 68, 0, 0, 0, 0, 0, 0}},
+    {96000, {64, 15, 187, 128, 0, 0, 0, 0, 0, 0}},
+    {176400, {64, 16, 172, 68, 0, 0, 0, 0, 0, 0}},
+    {192000, {64, 16, 187, 128, 0, 0, 0, 0, 0, 0}},
+    {352800, {64, 17, 172, 68, 0, 0, 0, 0, 0, 0}},
+    {2822400, {64, 20, 172, 68, 0, 0, 0, 0, 0, 0}},
+    {5644800, {64, 21, 172, 68, 0, 0, 0, 0, 0, 0}}
+};
+
+//=============================================================
+template <class T>
+AudioFile<T>::AudioFile()
+{
+    bitDepth = 16;
+    sampleRate = 44100;
+    samples.resize (1);
+    samples[0].resize (0);
+    audioFileFormat = AudioFileFormat::NotLoaded;
+}
+
+//=============================================================
+template <class T>
+uint32_t AudioFile<T>::getSampleRate() const
+{
+    return sampleRate;
+}
+
+//=============================================================
+template <class T>
+int AudioFile<T>::getNumChannels() const
+{
+    return (int)samples.size();
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::isMono() const
+{
+    return getNumChannels() == 1;
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::isStereo() const
+{
+    return getNumChannels() == 2;
+}
+
+//=============================================================
+template <class T>
+int AudioFile<T>::getBitDepth() const
+{
+    return bitDepth;
+}
+
+//=============================================================
+template <class T>
+int AudioFile<T>::getNumSamplesPerChannel() const
+{
+    if (samples.size() > 0)
+        return (int) samples[0].size();
+    else
+        return 0;
+}
+
+//=============================================================
+template <class T>
+double AudioFile<T>::getLengthInSeconds() const
+{
+    return (double)getNumSamplesPerChannel() / (double)sampleRate;
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::printSummary() const
+{
+    std::cout << "|======================================|" << std::endl;
+    std::cout << "Num Channels: " << getNumChannels() << std::endl;
+    std::cout << "Num Samples Per Channel: " << getNumSamplesPerChannel() << std::endl;
+    std::cout << "Sample Rate: " << sampleRate << std::endl;
+    std::cout << "Bit Depth: " << bitDepth << std::endl;
+    std::cout << "Length in Seconds: " << getLengthInSeconds() << std::endl;
+    std::cout << "Audio format: " << audioFormat << std::endl;
+    std::cout << "|======================================|" << std::endl;
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::setAudioBuffer (const AudioBuffer& newBuffer)
+{
+    int numChannels = (int)newBuffer.size();
+    
+    if (numChannels <= 0)
+    {
+        assert (false && "The buffer your are trying to use has no channels");
+        return false;
+    }
+    
+    unsigned int numSamples = newBuffer[0].size();
+    
+    // set the number of channels
+    samples.resize (newBuffer.size());
+    
+    for (int k = 0; k < getNumChannels(); k++)
+    {
+        assert (newBuffer[k].size() == numSamples);
+        
+        samples[k].resize (numSamples);
+        
+        for (unsigned int i = 0; i < numSamples; i++)
+        {
+            samples[k][i] = newBuffer[k][i];
+        }
+    }
+    
+    return true;
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::setAudioBufferSize (int numChannels, int numSamples)
+{
+    samples.resize (numChannels);
+    setNumSamplesPerChannel (numSamples);
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::setNumSamplesPerChannel (int numSamples)
+{
+    int originalSize = getNumSamplesPerChannel();
+    
+    for (int i = 0; i < getNumChannels();i++)
+    {
+        samples[i].resize (numSamples);
+        
+        // set any new samples to zero
+        if (numSamples > originalSize)
+            std::fill (samples[i].begin() + originalSize, samples[i].end(), (T)0.);
+    }
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::setNumChannels (int numChannels)
+{
+    int originalNumChannels = getNumChannels();
+    int originalNumSamplesPerChannel = getNumSamplesPerChannel();
+    
+    samples.resize (numChannels);
+    
+    // make sure any new channels are set to the right size
+    // and filled with zeros
+    if (numChannels > originalNumChannels)
+    {
+        for (int i = originalNumChannels; i < numChannels; i++)
+        {
+            samples[i].resize (originalNumSamplesPerChannel);
+            std::fill (samples[i].begin(), samples[i].end(), (T)0.);
+        }
+    }
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::setBitDepth (int numBitsPerSample)
+{
+    bitDepth = numBitsPerSample;
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::setSampleRate (uint32_t newSampleRate)
+{
+    sampleRate = newSampleRate;
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::load (std::string filePath)
+{
+    std::ifstream file (filePath, std::ios::binary);
+    
+    // check the file exists
+    if (! file.good())
+    {
+        std::cout << "ERROR: File doesn't exist or otherwise can't load file" << std::endl;
+        std::cout << filePath << std::endl;
+        return false;
+    }
+    
+    file.unsetf (std::ios::skipws);
+    std::istream_iterator<uint8_t> begin (file), end;
+    std::vector<uint8_t> fileData (begin, end);
+    
+    // get audio file format
+    audioFileFormat = determineAudioFileFormat (fileData);
+    
+    if (audioFileFormat == AudioFileFormat::Wave)
+    {
+        return decodeWaveFile (fileData);
+    }
+    else if (audioFileFormat == AudioFileFormat::Aiff)
+    {
+        return decodeAiffFile (fileData);
+    }
+    else
+    {
+        std::cout << "Audio File Type: " << "Error" << std::endl;
+        return false;
+    }
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::decodeWaveFile (std::vector<uint8_t>& fileData)
+{
+    // -----------------------------------------------------------
+    // HEADER CHUNK
+    std::string headerChunkID (fileData.begin(), fileData.begin() + 4);
+    //int32_t fileSizeInBytes = fourBytesToInt (fileData, 4) + 8;
+    std::string format (fileData.begin() + 8, fileData.begin() + 12);
+    
+    // -----------------------------------------------------------
+    // try and find the start points of key chunks
+    int indexOfDataChunk = getIndexOfString (fileData, "data");
+    int indexOfFormatChunk = getIndexOfString (fileData, "fmt");
+    
+    // if we can't find the data or format chunks, or the IDs/formats don't seem to be as expected
+    // then it is unlikely we'll able to read this file, so abort
+    if (indexOfDataChunk == -1 || indexOfFormatChunk == -1 || headerChunkID != "RIFF" || format != "WAVE")
+    {
+        std::cout << "ERROR: this doesn't seem to be a valid .WAV file" << std::endl;
+        return false;
+    }
+    
+    // -----------------------------------------------------------
+    // FORMAT CHUNK
+    int f = indexOfFormatChunk;
+    std::string formatChunkID (fileData.begin() + f, fileData.begin() + f + 4);
+    //int32_t formatChunkSize = fourBytesToInt (fileData, f + 4);
+    audioFormat = twoBytesToInt (fileData, f + 8);
+    int16_t numChannels = twoBytesToInt (fileData, f + 10);
+    sampleRate = (uint32_t) fourBytesToInt (fileData, f + 12);
+    int32_t numBytesPerSecond = fourBytesToInt (fileData, f + 16);
+    int16_t numBytesPerBlock = twoBytesToInt (fileData, f + 20);
+    bitDepth = (int) twoBytesToInt (fileData, f + 22);
+    
+    int numBytesPerSample = bitDepth / 8;
+    
+    // check that the audio format is PCM
+    if (audioFormat != 1)
+    {
+        std::cout << "ERROR: this is a compressed .WAV file and this library does not support decoding them at present" << std::endl;
+        return false;
+    }
+    
+    // check the number of channels is mono or stereo
+    if (numChannels < 1 ||numChannels > 2)
+    {
+        std::cout << "ERROR: this WAV file seems to be neither mono nor stereo (perhaps multi-track, or corrupted?)" << std::endl;
+        return false;
+    }
+    
+    // check header data is consistent
+    if ((numBytesPerSecond != (numChannels * sampleRate * bitDepth) / 8) || (numBytesPerBlock != (numChannels * numBytesPerSample)))
+    {
+        std::cout << "ERROR: the header data in this WAV file seems to be inconsistent" << std::endl;
+        return false;
+    }
+    
+    // check bit depth is either 8, 16 or 24 bit
+    if (bitDepth != 8 && bitDepth != 16 && bitDepth != 24)
+    {
+        std::cout << "ERROR: this file has a bit depth that is not 8, 16 or 24 bits" << std::endl;
+        return false;
+    }
+    
+    // -----------------------------------------------------------
+    // DATA CHUNK
+    int d = indexOfDataChunk;
+    std::string dataChunkID (fileData.begin() + d, fileData.begin() + d + 4);
+    int32_t dataChunkSize = fourBytesToInt (fileData, d + 4);
+    
+    int numSamples = dataChunkSize / (numChannels * bitDepth / 8);
+    int samplesStartIndex = indexOfDataChunk + 8;
+    
+    clearAudioBuffer();
+    samples.resize (numChannels);
+    
+    for (int i = 0; i < numSamples; i++)
+    {
+        for (int channel = 0; channel < numChannels; channel++)
+        {
+            int sampleIndex = samplesStartIndex + (numBytesPerBlock * i) + channel * numBytesPerSample;
+            
+            if (bitDepth == 8)
+            {
+                int32_t sampleAsInt = (int32_t) fileData[sampleIndex];
+                T sample = (T)(sampleAsInt - 128) / (T)128.;
+                
+                samples[channel].push_back (sample);
+            }
+            else if (bitDepth == 16)
+            {
+                int16_t sampleAsInt = twoBytesToInt (fileData, sampleIndex);
+                T sample = sampleAsInt;
+
+                if (std::is_floating_point_v<T>) {  // automatically constexpr
+                    sample = sixteenBitIntToSample (sampleAsInt);
+                }
+
+                samples[channel].push_back (sample);
+            }
+            else if (bitDepth == 24)
+            {
+                int32_t sampleAsInt = 0;
+                sampleAsInt = (fileData[sampleIndex + 2] << 16) | (fileData[sampleIndex + 1] << 8) | fileData[sampleIndex];
+                
+                if (sampleAsInt & 0x800000) //  if the 24th bit is set, this is a negative number in 24-bit world
+                    sampleAsInt = sampleAsInt | ~0xFFFFFF; // so make sure sign is extended to the 32 bit float
+
+                T sample = (T)sampleAsInt / (T)8388608.;
+                samples[channel].push_back (sample);
+            }
+            else
+            {
+                assert (false);
+            }
+        }
+    }
+
+    return true;
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::decodeAiffFile (std::vector<uint8_t>& fileData)
+{
+    // -----------------------------------------------------------
+    // HEADER CHUNK
+    std::string headerChunkID (fileData.begin(), fileData.begin() + 4);
+    //int32_t fileSizeInBytes = fourBytesToInt (fileData, 4, Endianness::BigEndian) + 8;
+    std::string format (fileData.begin() + 8, fileData.begin() + 12);
+    
+    // -----------------------------------------------------------
+    // try and find the start points of key chunks
+    int indexOfCommChunk = getIndexOfString (fileData, "COMM");
+    int indexOfSoundDataChunk = getIndexOfString (fileData, "SSND");
+    
+    // if we can't find the data or format chunks, or the IDs/formats don't seem to be as expected
+    // then it is unlikely we'll able to read this file, so abort
+    if (indexOfSoundDataChunk == -1 || indexOfCommChunk == -1 || headerChunkID != "FORM" || format != "AIFF")
+    {
+        std::cout << "ERROR: this doesn't seem to be a valid AIFF file" << std::endl;
+        return false;
+    }
+
+    // -----------------------------------------------------------
+    // COMM CHUNK
+    int p = indexOfCommChunk;
+    std::string commChunkID (fileData.begin() + p, fileData.begin() + p + 4);
+    //int32_t commChunkSize = fourBytesToInt (fileData, p + 4, Endianness::BigEndian);
+    int16_t numChannels = twoBytesToInt (fileData, p + 8, Endianness::BigEndian);
+    int32_t numSamplesPerChannel = fourBytesToInt (fileData, p + 10, Endianness::BigEndian);
+    bitDepth = (int) twoBytesToInt (fileData, p + 14, Endianness::BigEndian);
+    sampleRate = getAiffSampleRate (fileData, p + 16);
+    
+    // check the sample rate was properly decoded
+    if (sampleRate == -1)
+    {
+        std::cout << "ERROR: this AIFF file has an unsupported sample rate" << std::endl;
+        return false;
+    }
+    
+    // check the number of channels is mono or stereo
+    if (numChannels < 1 ||numChannels > 2)
+    {
+        std::cout << "ERROR: this AIFF file seems to be neither mono nor stereo (perhaps multi-track, or corrupted?)" << std::endl;
+        return false;
+    }
+    
+    // check bit depth is either 8, 16 or 24 bit
+    if (bitDepth != 8 && bitDepth != 16 && bitDepth != 24)
+    {
+        std::cout << "ERROR: this file has a bit depth that is not 8, 16 or 24 bits" << std::endl;
+        return false;
+    }
+    
+    // -----------------------------------------------------------
+    // SSND CHUNK
+    int s = indexOfSoundDataChunk;
+    std::string soundDataChunkID (fileData.begin() + s, fileData.begin() + s + 4);
+    int32_t soundDataChunkSize = fourBytesToInt (fileData, s + 4, Endianness::BigEndian);
+    int32_t offset = fourBytesToInt (fileData, s + 8, Endianness::BigEndian);
+    //int32_t blockSize = fourBytesToInt (fileData, s + 12, Endianness::BigEndian);
+    
+    int numBytesPerSample = bitDepth / 8;
+    int numBytesPerFrame = numBytesPerSample * numChannels;
+    int totalNumAudioSampleBytes = numSamplesPerChannel * numBytesPerFrame;
+    int samplesStartIndex = s + 16 + (int)offset;
+        
+    // sanity check the data
+    if ((soundDataChunkSize - 8) != totalNumAudioSampleBytes || totalNumAudioSampleBytes > (fileData.size() - samplesStartIndex))
+    {
+        std::cout << "ERROR: the metadatafor this file doesn't seem right" << std::endl;
+        return false;
+    }
+    
+    clearAudioBuffer();
+    samples.resize (numChannels);
+    
+    for (int i = 0; i < numSamplesPerChannel; i++)
+    {
+        for (int channel = 0; channel < numChannels; channel++)
+        {
+            int sampleIndex = samplesStartIndex + (numBytesPerFrame * i) + channel * numBytesPerSample;
+            
+            if (bitDepth == 8)
+            {
+                int8_t sampleAsSigned8Bit = (int8_t)fileData[sampleIndex];
+                T sample = (T)sampleAsSigned8Bit / (T)128.;
+                samples[channel].push_back (sample);
+            }
+            else if (bitDepth == 16)
+            {
+                int16_t sampleAsInt = twoBytesToInt (fileData, sampleIndex, Endianness::BigEndian);
+                T sample = sixteenBitIntToSample (sampleAsInt);
+                samples[channel].push_back (sample);
+            }
+            else if (bitDepth == 24)
+            {
+                int32_t sampleAsInt = 0;
+                sampleAsInt = (fileData[sampleIndex] << 16) | (fileData[sampleIndex + 1] << 8) | fileData[sampleIndex + 2];
+                
+                if (sampleAsInt & 0x800000) //  if the 24th bit is set, this is a negative number in 24-bit world
+                    sampleAsInt = sampleAsInt | ~0xFFFFFF; // so make sure sign is extended to the 32 bit float
+                
+                T sample = (T)sampleAsInt / (T)8388608.;
+                samples[channel].push_back (sample);
+            }
+            else
+            {
+                assert (false);
+            }
+        }
+    }
+    
+    return true;
+}
+
+//=============================================================
+template <class T>
+uint32_t AudioFile<T>::getAiffSampleRate (std::vector<uint8_t>& fileData, int sampleRateStartIndex)
+{
+    for (auto it : aiffSampleRateTable)
+    {
+        if (tenByteMatch (fileData, sampleRateStartIndex, it.second, 0))
+            return it.first;
+    }
+    
+    return -1;
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::tenByteMatch (std::vector<uint8_t>& v1, int startIndex1, std::vector<uint8_t>& v2, int startIndex2)
+{
+    for (int i = 0; i < 10; i++)
+    {
+        if (v1[startIndex1 + i] != v2[startIndex2 + i])
+            return false;
+    }
+    
+    return true;
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::addSampleRateToAiffData (std::vector<uint8_t>& fileData, uint32_t sampleRate)
+{
+    if (aiffSampleRateTable.count (sampleRate) > 0)
+    {
+        for (int i = 0; i < 10; i++)
+            fileData.push_back (aiffSampleRateTable[sampleRate][i]);
+    }
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::save (std::string filePath, AudioFileFormat format)
+{
+    if (format == AudioFileFormat::Wave)
+    {
+        return saveToWaveFile (filePath);
+    }
+    else if (format == AudioFileFormat::Aiff)
+    {
+        return saveToAiffFile (filePath);
+    }
+    
+    return false;
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::saveToWaveFile (std::string filePath)
+{
+    std::vector<uint8_t> fileData;
+    
+    int32_t dataChunkSize = getNumSamplesPerChannel() * (getNumChannels() * bitDepth / 8);
+    
+    // -----------------------------------------------------------
+    // HEADER CHUNK
+    addStringToFileData (fileData, "RIFF");
+    
+    // The file size in bytes is the header chunk size (4, not counting RIFF and WAVE) + the format
+    // chunk size (24) + the metadata part of the data chunk plus the actual data chunk size
+    int32_t fileSizeInBytes = 4 + 24 + 8 + dataChunkSize;
+    addInt32ToFileData (fileData, fileSizeInBytes);
+    
+    addStringToFileData (fileData, "WAVE");
+    
+    // -----------------------------------------------------------
+    // FORMAT CHUNK
+    addStringToFileData (fileData, "fmt ");
+    addInt32ToFileData (fileData, 16); // format chunk size (16 for PCM)
+    addInt16ToFileData (fileData, 1); // audio format = 1
+    addInt16ToFileData (fileData, (int16_t)getNumChannels()); // num channels
+    addInt32ToFileData (fileData, (int32_t)sampleRate); // sample rate
+    
+    int32_t numBytesPerSecond = (int32_t) ((getNumChannels() * sampleRate * bitDepth) / 8);
+    addInt32ToFileData (fileData, numBytesPerSecond);
+    
+    int16_t numBytesPerBlock = getNumChannels() * (bitDepth / 8);
+    addInt16ToFileData (fileData, numBytesPerBlock);
+    
+    addInt16ToFileData (fileData, (int16_t)bitDepth);
+    
+    // -----------------------------------------------------------
+    // DATA CHUNK
+    addStringToFileData (fileData, "data");
+    addInt32ToFileData (fileData, dataChunkSize);
+    
+    for (int i = 0; i < getNumSamplesPerChannel(); i++)
+    {
+        for (int channel = 0; channel < getNumChannels(); channel++)
+        {
+            if (bitDepth == 8)
+            {
+                int32_t sampleAsInt = ((samples[channel][i] * (T)128.) + 128.);
+                uint8_t byte = (uint8_t)sampleAsInt;
+                fileData.push_back (byte);
+            }
+            else if (bitDepth == 16)
+            {
+                int16_t sampleAsInt = (int16_t) (samples[channel][i] * (T)32768.);
+                addInt16ToFileData (fileData, sampleAsInt);
+            }
+            else if (bitDepth == 24)
+            {
+                int32_t sampleAsIntAgain = (int32_t) (samples[channel][i] * (T)8388608.);
+                
+                uint8_t bytes[3];
+                bytes[2] = (uint8_t) (sampleAsIntAgain >> 16) & 0xFF;
+                bytes[1] = (uint8_t) (sampleAsIntAgain >>  8) & 0xFF;
+                bytes[0] = (uint8_t) sampleAsIntAgain & 0xFF;
+                
+                fileData.push_back (bytes[0]);
+                fileData.push_back (bytes[1]);
+                fileData.push_back (bytes[2]);
+            }
+            else
+            {
+                assert (false && "Trying to write a file with unsupported bit depth");
+                return false;
+            }
+        }
+    }
+    
+    // check that the various sizes we put in the metadata are correct
+    if (fileSizeInBytes != (fileData.size() - 8) || dataChunkSize != (getNumSamplesPerChannel() * getNumChannels() * (bitDepth / 8)))
+    {
+        std::cout << "ERROR: couldn't save file to " << filePath << std::endl;
+        return false;
+    }
+    
+    // try to write the file
+    return writeDataToFile (fileData, filePath);
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::saveToAiffFile (std::string filePath)
+{
+    std::vector<uint8_t> fileData;
+    
+    int32_t numBytesPerSample = bitDepth / 8;
+    int32_t numBytesPerFrame = numBytesPerSample * getNumChannels();
+    int32_t totalNumAudioSampleBytes = getNumSamplesPerChannel() * numBytesPerFrame;
+    int32_t soundDataChunkSize = totalNumAudioSampleBytes + 8;
+    
+    // -----------------------------------------------------------
+    // HEADER CHUNK
+    addStringToFileData (fileData, "FORM");
+    
+    // The file size in bytes is the header chunk size (4, not counting FORM and AIFF) + the COMM
+    // chunk size (26) + the metadata part of the SSND chunk plus the actual data chunk size
+    int32_t fileSizeInBytes = 4 + 26 + 16 + totalNumAudioSampleBytes;
+    addInt32ToFileData (fileData, fileSizeInBytes, Endianness::BigEndian);
+    
+    addStringToFileData (fileData, "AIFF");
+    
+    // -----------------------------------------------------------
+    // COMM CHUNK
+    addStringToFileData (fileData, "COMM");
+    addInt32ToFileData (fileData, 18, Endianness::BigEndian); // commChunkSize
+    addInt16ToFileData (fileData, getNumChannels(), Endianness::BigEndian); // num channels
+    addInt32ToFileData (fileData, getNumSamplesPerChannel(), Endianness::BigEndian); // num samples per channel
+    addInt16ToFileData (fileData, bitDepth, Endianness::BigEndian); // bit depth
+    addSampleRateToAiffData (fileData, sampleRate);
+    
+    // -----------------------------------------------------------
+    // SSND CHUNK
+    addStringToFileData (fileData, "SSND");
+    addInt32ToFileData (fileData, soundDataChunkSize, Endianness::BigEndian);
+    addInt32ToFileData (fileData, 0, Endianness::BigEndian); // offset
+    addInt32ToFileData (fileData, 0, Endianness::BigEndian); // block size
+    
+    for (int i = 0; i < getNumSamplesPerChannel(); i++)
+    {
+        for (int channel = 0; channel < getNumChannels(); channel++)
+        {
+            if (bitDepth == 8)
+            {
+                int32_t sampleAsInt = (int32_t)(samples[channel][i] * (T)128.);
+                uint8_t byte = (uint8_t)sampleAsInt;
+                fileData.push_back (byte);
+            }
+            else if (bitDepth == 16)
+            {
+                int16_t sampleAsInt = (int16_t) (samples[channel][i] * (T)32768.);
+                addInt16ToFileData (fileData, sampleAsInt, Endianness::BigEndian);
+            }
+            else if (bitDepth == 24)
+            {
+                int32_t sampleAsIntAgain = (int32_t) (samples[channel][i] * (T)8388608.);
+                
+                uint8_t bytes[3];
+                bytes[0] = (uint8_t) (sampleAsIntAgain >> 16) & 0xFF;
+                bytes[1] = (uint8_t) (sampleAsIntAgain >>  8) & 0xFF;
+                bytes[2] = (uint8_t) sampleAsIntAgain & 0xFF;
+                
+                fileData.push_back (bytes[0]);
+                fileData.push_back (bytes[1]);
+                fileData.push_back (bytes[2]);
+            }
+            else
+            {
+                assert (false && "Trying to write a file with unsupported bit depth");
+                return false;
+            }
+        }
+    }
+    
+    // check that the various sizes we put in the metadata are correct
+    if (fileSizeInBytes != (fileData.size() - 8) || soundDataChunkSize != getNumSamplesPerChannel() *  numBytesPerFrame + 8)
+    {
+        std::cout << "ERROR: couldn't save file to " << filePath << std::endl;
+        return false;
+    }
+    
+    // try to write the file
+    return writeDataToFile (fileData, filePath);
+}
+
+//=============================================================
+template <class T>
+bool AudioFile<T>::writeDataToFile (std::vector<uint8_t>& fileData, std::string filePath)
+{
+    std::ofstream outputFile (filePath, std::ios::binary);
+    
+    if (outputFile.is_open())
+    {
+        for (int i = 0; i < fileData.size(); i++)
+        {
+            char value = (char) fileData[i];
+            outputFile.write (&value, sizeof (char));
+        }
+        
+        outputFile.close();
+        
+        return true;
+    }
+    
+    return false;
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::addStringToFileData (std::vector<uint8_t>& fileData, std::string s)
+{
+    for (int i = 0; i < s.length();i++)
+        fileData.push_back ((uint8_t) s[i]);
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::addInt32ToFileData (std::vector<uint8_t>& fileData, int32_t i, Endianness endianness)
+{
+    uint8_t bytes[4];
+    
+    if (endianness == Endianness::LittleEndian)
+    {
+        bytes[3] = (i >> 24) & 0xFF;
+        bytes[2] = (i >> 16) & 0xFF;
+        bytes[1] = (i >> 8) & 0xFF;
+        bytes[0] = i & 0xFF;
+    }
+    else
+    {
+        bytes[0] = (i >> 24) & 0xFF;
+        bytes[1] = (i >> 16) & 0xFF;
+        bytes[2] = (i >> 8) & 0xFF;
+        bytes[3] = i & 0xFF;
+    }
+    
+    for (int i = 0; i < 4; i++)
+        fileData.push_back (bytes[i]);
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::addInt16ToFileData (std::vector<uint8_t>& fileData, int16_t i, Endianness endianness)
+{
+    uint8_t bytes[2];
+    
+    if (endianness == Endianness::LittleEndian)
+    {
+        bytes[1] = (i >> 8) & 0xFF;
+        bytes[0] = i & 0xFF;
+    }
+    else
+    {
+        bytes[0] = (i >> 8) & 0xFF;
+        bytes[1] = i & 0xFF;
+    }
+    
+    fileData.push_back (bytes[0]);
+    fileData.push_back (bytes[1]);
+}
+
+//=============================================================
+template <class T>
+void AudioFile<T>::clearAudioBuffer()
+{
+    for (int i = 0; i < samples.size();i++)
+    {
+        samples[i].clear();
+    }
+    
+    samples.clear();
+}
+
+//=============================================================
+template <class T>
+AudioFileFormat AudioFile<T>::determineAudioFileFormat (std::vector<uint8_t>& fileData)
+{
+    std::string header (fileData.begin(), fileData.begin() + 4);
+    
+    if (header == "RIFF")
+        return AudioFileFormat::Wave;
+    else if (header == "FORM")
+        return AudioFileFormat::Aiff;
+    else
+        return AudioFileFormat::Error;
+}
+
+//=============================================================
+template <class T>
+int32_t AudioFile<T>::fourBytesToInt (std::vector<uint8_t>& source, int startIndex, Endianness endianness)
+{
+    int32_t result;
+    
+    if (endianness == Endianness::LittleEndian)
+        result = (source[startIndex + 3] << 24) | (source[startIndex + 2] << 16) | (source[startIndex + 1] << 8) | source[startIndex];
+    else
+        result = (source[startIndex] << 24) | (source[startIndex + 1] << 16) | (source[startIndex + 2] << 8) | source[startIndex + 3];
+    
+    return result;
+}
+
+//=============================================================
+template <class T>
+int16_t AudioFile<T>::twoBytesToInt (std::vector<uint8_t>& source, int startIndex, Endianness endianness)
+{
+    int16_t result;
+    
+    if (endianness == Endianness::LittleEndian)
+        result = (source[startIndex + 1] << 8) | source[startIndex];
+    else
+        result = (source[startIndex] << 8) | source[startIndex + 1];
+    
+    return result;
+}
+
+//=============================================================
+template <class T>
+int AudioFile<T>::getIndexOfString (std::vector<uint8_t>& source, std::string stringToSearchFor)
+{
+    unsigned int index = -1;
+    int stringLength = (int)stringToSearchFor.length();
+    
+    for (int i = 0; i < source.size() - stringLength;i++)
+    {
+        std::string section (source.begin() + i, source.begin() + i + stringLength);
+        
+        if (section == stringToSearchFor)
+        {
+            index = i;
+            break;
+        }
+    }
+    
+    return index;
+}
+
+//=============================================================
+template <class T>
+T AudioFile<T>::sixteenBitIntToSample (int16_t sample)
+{
+    return (T)sample / (T)32768.;
+}
+
+//===========================================================
+template class AudioFile<float>;
+template class AudioFile<double>;
+template class AudioFile<int16_t>;
\ No newline at end of file
diff --git a/story-editor/tools/audio/AudioFile.h b/story-editor/tools/audio/AudioFile.h
new file mode 100755
index 0000000..9effd21
--- /dev/null
+++ b/story-editor/tools/audio/AudioFile.h
@@ -0,0 +1,173 @@
+//=======================================================================
+/** @file AudioFile.h
+ *  @author Adam Stark
+ *  @copyright Copyright (C) 2017  Adam Stark
+ *
+ * This file is part of the 'AudioFile' library
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+//=======================================================================
+
+#ifndef _AS_AudioFile_h
+#define _AS_AudioFile_h
+
+#include <iostream>
+#include <vector>
+#include <assert.h>
+#include <string>
+
+
+//=============================================================
+/** The different types of audio file, plus some other types to 
+ * indicate a failure to load a file, or that one hasn't been
+ * loaded yet
+ */
+enum class AudioFileFormat
+{
+    Error,
+    NotLoaded,
+    Wave,
+    Aiff
+};
+
+//=============================================================
+template <class T>
+class AudioFile
+{
+public:
+    
+    //=============================================================
+    typedef std::vector<std::vector<T> > AudioBuffer;
+    
+    //=============================================================
+    /** Constructor */
+    AudioFile();
+        
+    //=============================================================
+    /** Loads an audio file from a given file path.
+     * @Returns true if the file was successfully loaded
+     */
+    bool load (std::string filePath);
+    
+    /** Saves an audio file to a given file path.
+     * @Returns true if the file was successfully saved
+     */
+    bool save (std::string filePath, AudioFileFormat format = AudioFileFormat::Wave);
+        
+    //=============================================================
+    /** @Returns the sample rate */
+    uint32_t getSampleRate() const;
+    
+    /** @Returns the number of audio channels in the buffer */
+    int getNumChannels() const;
+
+    /** @Returns true if the audio file is mono */
+    bool isMono() const;
+    
+    /** @Returns true if the audio file is stereo */
+    bool isStereo() const;
+    
+    /** @Returns the bit depth of each sample */
+    int getBitDepth() const;
+    
+    /** @Returns the number of samples per channel */
+    int getNumSamplesPerChannel() const;
+    
+    /** @Returns the length in seconds of the audio file based on the number of samples and sample rate */
+    double getLengthInSeconds() const;
+    
+    /** Prints a summary of the audio file to the console */
+    void printSummary() const;
+    
+    //=============================================================
+    
+    /** Set the audio buffer for this AudioFile by copying samples from another buffer.
+     * @Returns true if the buffer was copied successfully.
+     */
+    bool setAudioBuffer (const AudioBuffer &newBuffer);
+    
+    /** Sets the audio buffer to a given number of channels and number of samples per channel. This will try to preserve
+     * the existing audio, adding zeros to any new channels or new samples in a given channel.
+     */
+    void setAudioBufferSize (int numChannels, int numSamples);
+    
+    /** Sets the number of samples per channel in the audio buffer. This will try to preserve
+     * the existing audio, adding zeros to new samples in a given channel if the number of samples is increased.
+     */
+    void setNumSamplesPerChannel (int numSamples);
+    
+    /** Sets the number of channels. New channels will have the correct number of samples and be initialised to zero */
+    void setNumChannels (int numChannels);
+    
+    /** Sets the bit depth for the audio file. If you use the save() function, this bit depth rate will be used */
+    void setBitDepth (int numBitsPerSample);
+    
+    /** Sets the sample rate for the audio file. If you use the save() function, this sample rate will be used */
+    void setSampleRate (uint32_t newSampleRate);
+    
+    //=============================================================
+    /** A vector of vectors holding the audio samples for the AudioFile. You can 
+     * access the samples by channel and then by sample index, i.e:
+     *
+     *      samples[channel][sampleIndex]
+     */
+    AudioBuffer samples;
+    
+private:
+    
+    //=============================================================
+    enum class Endianness
+    {
+        LittleEndian,
+        BigEndian
+    };
+    
+    //=============================================================
+    AudioFileFormat determineAudioFileFormat (std::vector<uint8_t>& fileData);
+    bool decodeWaveFile (std::vector<uint8_t>& fileData);
+    bool decodeAiffFile (std::vector<uint8_t>& fileData);
+    
+    //=============================================================
+    bool saveToWaveFile (std::string filePath);
+    bool saveToAiffFile (std::string filePath);
+    
+    //=============================================================
+    void clearAudioBuffer();
+    
+    //=============================================================
+    int32_t fourBytesToInt (std::vector<uint8_t>& source, int startIndex, Endianness endianness = Endianness::LittleEndian);
+    int16_t twoBytesToInt (std::vector<uint8_t>& source, int startIndex, Endianness endianness = Endianness::LittleEndian);
+    int getIndexOfString (std::vector<uint8_t>& source, std::string s);
+    T sixteenBitIntToSample (int16_t sample);
+    uint32_t getAiffSampleRate (std::vector<uint8_t>& fileData, int sampleRateStartIndex);
+    bool tenByteMatch (std::vector<uint8_t>& v1, int startIndex1, std::vector<uint8_t>& v2, int startIndex2);
+    void addSampleRateToAiffData (std::vector<uint8_t>& fileData, uint32_t sampleRate);
+    
+    //=============================================================
+    void addStringToFileData (std::vector<uint8_t>& fileData, std::string s);
+    void addInt32ToFileData (std::vector<uint8_t>& fileData, int32_t i, Endianness endianness = Endianness::LittleEndian);
+    void addInt16ToFileData (std::vector<uint8_t>& fileData, int16_t i, Endianness endianness = Endianness::LittleEndian);
+    
+    //=============================================================
+    bool writeDataToFile (std::vector<uint8_t>& fileData, std::string filePath);
+    
+    //=============================================================
+    AudioFileFormat audioFileFormat;
+    int16_t audioFormat{-1};
+    uint32_t sampleRate;
+    int bitDepth;
+};
+
+#endif /* AudioFile_h */
diff --git a/story-editor/tools/audio/CMakeLists.txt b/story-editor/tools/audio/CMakeLists.txt
new file mode 100644
index 0000000..9f47062
--- /dev/null
+++ b/story-editor/tools/audio/CMakeLists.txt
@@ -0,0 +1,6 @@
+cmake_minimum_required(VERSION 3.24)
+project(audio C CXX)
+
+set(CMAKE_CXX_STANDARD 17)
+
+add_executable(audio main.cpp AudioFile.cpp AudioFile.h)
diff --git a/story-editor/tools/audio/main.cpp b/story-editor/tools/audio/main.cpp
new file mode 100755
index 0000000..ed785a0
--- /dev/null
+++ b/story-editor/tools/audio/main.cpp
@@ -0,0 +1,86 @@
+
+
+#include <vector>
+#include <iostream>
+#include <string>
+#include <cstdint>
+
+#include "AudioFile.h"
+
+std::string GetSuffix(const std::string &path)
+{
+    return path.substr(path.find_last_of(".") + 1);
+}
+
+std::string GetFullBaseName(const std::string &path)
+{
+    std::string suffix = GetSuffix(path);
+    std::string basename = path;
+    basename.erase(path.size()-(suffix.size()+1));
+    return basename;
+}
+
+void PrintHelp(const char *exeName)
+{
+    std::cout << exeName << " - WAV from/to SVG converter" << std::endl;
+    std::cout << "Usage: " << exeName << " myfile.[wav|svg]" << std::endl;
+}
+
+
+void WavInspector(const std::string &inputFileName)
+{
+
+    AudioFile<int16_t> audioFile;
+    bool loadedOK = audioFile.load(inputFileName);
+
+    if (loadedOK)
+    {
+        int samples = audioFile.getNumSamplesPerChannel();
+
+
+        audioFile.printSummary();
+        for (int k = 0; k < audioFile.getNumChannels(); k++)
+        {
+            std::cout << "Found channel: " << k << std::endl;
+            for (int i = 0; i < 10; i++)
+            {
+                int16_t sampleRaw = audioFile.samples[k][i];
+                std::cout << (int)sampleRaw << std::endl;
+            }
+        }
+
+    }
+}
+
+
+int main(int argc, char *argv[])
+{
+    int ret = 0;
+    if (argc > 1)
+    {
+        std::cout << GetFullBaseName(argv[1]) << std::endl;
+
+        std::string baseName = GetFullBaseName(argv[1]);
+        std::string suffix = GetSuffix(argv[1]);
+
+        if(suffix == "wav")
+        {
+            WavInspector(argv[1]);
+        }
+        else
+        {
+            ret = -1;
+        }
+    }
+    else
+    {
+        ret = -2;
+    }
+
+    if (ret != 0)
+    {
+        PrintHelp(argv[0]);
+    }
+
+    return ret;
+}