//=======================================================================
/** @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 <cassert>
#include <string>
#include <cstring>
#include <fstream>
#include <unordered_map>
#include <iterator>
#include <algorithm>
// disable some warnings on Windows
#if defined (_MSC_VER)
__pragma(warning (push))
__pragma(warning (disable : 4244))
__pragma(warning (disable : 4457))
__pragma(warning (disable : 4458))
__pragma(warning (disable : 4389))
__pragma(warning (disable : 4996))
#elif defined (__GNUC__)
_Pragma("GCC diagnostic push")
_Pragma("GCC diagnostic ignored \"-Wconversion\"")
_Pragma("GCC diagnostic ignored \"-Wsign-compare\"")
_Pragma("GCC diagnostic ignored \"-Wshadow\"")
#endif
//=============================================================
/** 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();
/** Constructor, using a given file path to load a file */
AudioFile (std::string filePath);
//=============================================================
/** 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 (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);
//=============================================================
/** Sets whether the library should log error messages to the console. By default this is true */
void shouldLogErrorsToConsole (bool logErrors);
//=============================================================
/** 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;
//=============================================================
/** An optional iXML chunk that can be added to the AudioFile.
*/
std::string iXMLChunk;
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);
int getIndexOfChunk (std::vector<uint8_t>& source, const std::string& chunkHeaderID, int startIndex, Endianness endianness = Endianness::LittleEndian);
//=============================================================
T sixteenBitIntToSample (int16_t sample);
int16_t sampleToSixteenBitInt (T sample);
//=============================================================
uint8_t sampleToSingleByte (T sample);
T singleByteToSample (uint8_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);
T clamp (T v1, T minValue, T maxValue);
//=============================================================
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);
//=============================================================
void reportError (std::string errorMessage);
//=============================================================
AudioFileFormat audioFileFormat;
uint32_t sampleRate;
int bitDepth;
bool logErrorsToConsole {true};
};
//=============================================================
// Pre-defined 10-byte representations of common sample rates
static 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}}
};
//=============================================================
enum WavAudioFormat
{
PCM = 0x0001,
IEEEFloat = 0x0003,
ALaw = 0x0006,
MULaw = 0x0007,
Extensible = 0xFFFE
};
//=============================================================
enum AIFFAudioFormat
{
Uncompressed,
Compressed,
Error
};
//=============================================================
/* IMPLEMENTATION */
//=============================================================
//=============================================================
template <class T>
AudioFile<T>::AudioFile()
{
static_assert(std::is_floating_point<T>::value, "ERROR: This version of AudioFile only supports floating point sample formats");
bitDepth = 16;
sampleRate = 44100;
samples.resize (1);
samples[0].resize (0);
audioFileFormat = AudioFileFormat::NotLoaded;
}
//=============================================================
template <class T>
AudioFile<T>::AudioFile (std::string filePath)
: AudioFile<T>()
{
load (filePath);
}
//=============================================================
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 << "|======================================|" << std::endl;
}
//=============================================================
template <class T>
bool AudioFile<T>::setAudioBuffer (AudioBuffer& newBuffer)
{
int numChannels = (int)newBuffer.size();
if (numChannels <= 0)
{
assert (false && "The buffer your are trying to use has no channels");
return false;
}
size_t 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 (size_t 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>
void AudioFile<T>::shouldLogErrorsToConsole (bool logErrors)
{
logErrorsToConsole = logErrors;
}
//=============================================================
template <class T>
bool AudioFile<T>::load (std::string filePath)
{
std::ifstream file (filePath, std::ios::binary);
// check the file exists
if (! file.good())
{
reportError ("ERROR: File doesn't exist or otherwise can't load file\n" + filePath);
return false;
}
std::vector<uint8_t> fileData;
file.unsetf (std::ios::skipws);
file.seekg (0, std::ios::end);
size_t length = file.tellg();
file.seekg (0, std::ios::beg);
// allocate
fileData.resize (length);
file.read(reinterpret_cast<char*> (fileData.data()), length);
file.close();
if (file.gcount() != length)
{
reportError ("ERROR: Couldn't read entire file\n" + filePath);
return false;
}
// get audio file format
audioFileFormat = determineAudioFileFormat (fileData);
if (audioFileFormat == AudioFileFormat::Wave)
{
return decodeWaveFile (fileData);
}
else if (audioFileFormat == AudioFileFormat::Aiff)
{
return decodeAiffFile (fileData);
}
else
{
reportError ("Audio File Type: Error");
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 = getIndexOfChunk (fileData, "data", 12);
int indexOfFormatChunk = getIndexOfChunk (fileData, "fmt ", 12);
int indexOfXMLChunk = getIndexOfChunk (fileData, "iXML", 12);
// 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")
{
reportError ("ERROR: this doesn't seem to be a valid .WAV file");
return false;
}
// -----------------------------------------------------------
// FORMAT CHUNK
int f = indexOfFormatChunk;
std::string formatChunkID (fileData.begin() + f, fileData.begin() + f + 4);
//int32_t formatChunkSize = fourBytesToInt (fileData, f + 4);
uint16_t audioFormat = twoBytesToInt (fileData, f + 8);
uint16_t numChannels = twoBytesToInt (fileData, f + 10);
sampleRate = (uint32_t) fourBytesToInt (fileData, f + 12);
uint32_t numBytesPerSecond = fourBytesToInt (fileData, f + 16);
uint16_t numBytesPerBlock = twoBytesToInt (fileData, f + 20);
bitDepth = (int) twoBytesToInt (fileData, f + 22);
uint16_t numBytesPerSample = static_cast<uint16_t> (bitDepth) / 8;
// check that the audio format is PCM or Float or extensible
if (audioFormat != WavAudioFormat::PCM && audioFormat != WavAudioFormat::IEEEFloat && audioFormat != WavAudioFormat::Extensible)
{
reportError ("ERROR: this .WAV file is encoded in a format that this library does not support at present");
return false;
}
// check the number of channels is mono or stereo
if (numChannels < 1 || numChannels > 128)
{
reportError ("ERROR: this WAV file seems to be an invalid number of channels (or corrupted?)");
return false;
}
// check header data is consistent
if (numBytesPerSecond != static_cast<uint32_t> ((numChannels * sampleRate * bitDepth) / 8) || numBytesPerBlock != (numChannels * numBytesPerSample))
{
reportError ("ERROR: the header data in this WAV file seems to be inconsistent");
return false;
}
// check bit depth is either 8, 16, 24 or 32 bit
if (bitDepth != 8 && bitDepth != 16 && bitDepth != 24 && bitDepth != 32)
{
reportError ("ERROR: this file has a bit depth that is not 8, 16, 24 or 32 bits");
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 ((sampleIndex + (bitDepth / 8) - 1) >= fileData.size())
{
reportError ("ERROR: read file error as the metadata indicates more samples than there are in the file data");
return false;
}
if (bitDepth == 8)
{
T sample = singleByteToSample (fileData[sampleIndex]);
samples[channel].push_back (sample);
}
else if (bitDepth == 16)
{
int16_t sampleAsInt = twoBytesToInt (fileData, sampleIndex);
T 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 if (bitDepth == 32)
{
int32_t sampleAsInt = fourBytesToInt (fileData, sampleIndex);
T sample;
if (audioFormat == WavAudioFormat::IEEEFloat)
sample = (T)reinterpret_cast<float&> (sampleAsInt);
else // assume PCM
sample = (T) sampleAsInt / static_cast<float> (std::numeric_limits<std::int32_t>::max());
samples[channel].push_back (sample);
}
else
{
assert (false);
}
}
}
// -----------------------------------------------------------
// iXML CHUNK
if (indexOfXMLChunk != -1)
{
int32_t chunkSize = fourBytesToInt (fileData, indexOfXMLChunk + 4);
iXMLChunk = std::string ((const char*) &fileData[indexOfXMLChunk + 8], chunkSize);
}
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);
int audioFormat = format == "AIFF" ? AIFFAudioFormat::Uncompressed : format == "AIFC" ? AIFFAudioFormat::Compressed : AIFFAudioFormat::Error;
// -----------------------------------------------------------
// try and find the start points of key chunks
int indexOfCommChunk = getIndexOfChunk (fileData, "COMM", 12, Endianness::BigEndian);
int indexOfSoundDataChunk = getIndexOfChunk (fileData, "SSND", 12, Endianness::BigEndian);
int indexOfXMLChunk = getIndexOfChunk (fileData, "iXML", 12, Endianness::BigEndian);
// 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" || audioFormat == AIFFAudioFormat::Error)
{
reportError ("ERROR: this doesn't seem to be a valid AIFF file");
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 == 0)
{
reportError ("ERROR: this AIFF file has an unsupported sample rate");
return false;
}
// check the number of channels is mono or stereo
if (numChannels < 1 ||numChannels > 2)
{
reportError ("ERROR: this AIFF file seems to be neither mono nor stereo (perhaps multi-track, or corrupted?)");
return false;
}
// check bit depth is either 8, 16, 24 or 32-bit
if (bitDepth != 8 && bitDepth != 16 && bitDepth != 24 && bitDepth != 32)
{
reportError ("ERROR: this file has a bit depth that is not 8, 16, 24 or 32 bits");
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 > static_cast<long>(fileData.size() - samplesStartIndex))
{
reportError ("ERROR: the metadatafor this file doesn't seem right");
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 ((sampleIndex + (bitDepth / 8) - 1) >= fileData.size())
{
reportError ("ERROR: read file error as the metadata indicates more samples than there are in the file data");
return false;
}
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 if (bitDepth == 32)
{
int32_t sampleAsInt = fourBytesToInt (fileData, sampleIndex, Endianness::BigEndian);
T sample;
if (audioFormat == AIFFAudioFormat::Compressed)
sample = (T)reinterpret_cast<float&> (sampleAsInt);
else // assume uncompressed
sample = (T) sampleAsInt / static_cast<float> (std::numeric_limits<std::int32_t>::max());
samples[channel].push_back (sample);
}
else
{
assert (false);
}
}
}
// -----------------------------------------------------------
// iXML CHUNK
if (indexOfXMLChunk != -1)
{
int32_t chunkSize = fourBytesToInt (fileData, indexOfXMLChunk + 4);
iXMLChunk = std::string ((const char*) &fileData[indexOfXMLChunk + 8], chunkSize);
}
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 0;
}
//=============================================================
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);
int16_t audioFormat = bitDepth == 32 ? WavAudioFormat::IEEEFloat : WavAudioFormat::PCM;
int32_t formatChunkSize = audioFormat == WavAudioFormat::PCM ? 16 : 18;
int32_t iXMLChunkSize = static_cast<int32_t> (iXMLChunk.size());
// -----------------------------------------------------------
// 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 + formatChunkSize + 8 + 8 + dataChunkSize;
if (iXMLChunkSize > 0)
{
fileSizeInBytes += (8 + iXMLChunkSize);
}
addInt32ToFileData (fileData, fileSizeInBytes);
addStringToFileData (fileData, "WAVE");
// -----------------------------------------------------------
// FORMAT CHUNK
addStringToFileData (fileData, "fmt ");
addInt32ToFileData (fileData, formatChunkSize); // format chunk size (16 for PCM)
addInt16ToFileData (fileData, audioFormat); // audio format
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);
if (audioFormat == WavAudioFormat::IEEEFloat)
addInt16ToFileData (fileData, 0); // extension size
// -----------------------------------------------------------
// 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)
{
uint8_t byte = sampleToSingleByte (samples[channel][i]);
fileData.push_back (byte);
}
else if (bitDepth == 16)
{
int16_t sampleAsInt = sampleToSixteenBitInt (samples[channel][i]);
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 if (bitDepth == 32)
{
int32_t sampleAsInt;
if (audioFormat == WavAudioFormat::IEEEFloat)
sampleAsInt = (int32_t) reinterpret_cast<int32_t&> (samples[channel][i]);
else // assume PCM
sampleAsInt = (int32_t) (samples[channel][i] * std::numeric_limits<int32_t>::max());
addInt32ToFileData (fileData, sampleAsInt, Endianness::LittleEndian);
}
else
{
assert (false && "Trying to write a file with unsupported bit depth");
return false;
}
}
}
// -----------------------------------------------------------
// iXML CHUNK
if (iXMLChunkSize > 0)
{
addStringToFileData (fileData, "iXML");
addInt32ToFileData (fileData, iXMLChunkSize);
addStringToFileData (fileData, iXMLChunk);
}
// check that the various sizes we put in the metadata are correct
if (fileSizeInBytes != static_cast<int32_t> (fileData.size() - 8) || dataChunkSize != (getNumSamplesPerChannel() * getNumChannels() * (bitDepth / 8)))
{
reportError ("ERROR: couldn't save file to " + filePath);
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;
int32_t iXMLChunkSize = static_cast<int32_t> (iXMLChunk.size());
// -----------------------------------------------------------
// 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;
if (iXMLChunkSize > 0)
{
fileSizeInBytes += (8 + iXMLChunkSize);
}
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)
{
uint8_t byte = sampleToSingleByte (samples[channel][i]);
fileData.push_back (byte);
}
else if (bitDepth == 16)
{
int16_t sampleAsInt = sampleToSixteenBitInt (samples[channel][i]);
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 if (bitDepth == 32)
{
// write samples as signed integers (no implementation yet for floating point, but looking at WAV implementation should help)
int32_t sampleAsInt = (int32_t) (samples[channel][i] * std::numeric_limits<int32_t>::max());
addInt32ToFileData (fileData, sampleAsInt, Endianness::BigEndian);
}
else
{
assert (false && "Trying to write a file with unsupported bit depth");
return false;
}
}
}
// -----------------------------------------------------------
// iXML CHUNK
if (iXMLChunkSize > 0)
{
addStringToFileData (fileData, "iXML");
addInt32ToFileData (fileData, iXMLChunkSize, Endianness::BigEndian);
addStringToFileData (fileData, iXMLChunk);
}
// check that the various sizes we put in the metadata are correct
if (fileSizeInBytes != static_cast<int32_t> (fileData.size() - 8) || soundDataChunkSize != getNumSamplesPerChannel() * numBytesPerFrame + 8)
{
reportError ("ERROR: couldn't save file to " + filePath);
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 (size_t 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 (size_t 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 (size_t 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)
{
int index = -1;
int stringLength = (int)stringToSearchFor.length();
for (size_t i = 0; i < source.size() - stringLength;i++)
{
std::string section (source.begin() + i, source.begin() + i + stringLength);
if (section == stringToSearchFor)
{
index = static_cast<int> (i);
break;
}
}
return index;
}
//=============================================================
template <class T>
int AudioFile<T>::getIndexOfChunk (std::vector<uint8_t>& source, const std::string& chunkHeaderID, int startIndex, Endianness endianness)
{
constexpr int dataLen = 4;
if (chunkHeaderID.size() != dataLen)
{
assert (false && "Invalid chunk header ID string");
return -1;
}
int i = startIndex;
while (i < source.size() - dataLen)
{
if (memcmp (&source[i], chunkHeaderID.data(), dataLen) == 0)
{
return i;
}
i += dataLen;
auto chunkSize = fourBytesToInt (source, i, endianness);
i += (dataLen + chunkSize);
}
return -1;
}
//=============================================================
template <class T>
T AudioFile<T>::sixteenBitIntToSample (int16_t sample)
{
return static_cast<T> (sample) / static_cast<T> (32768.);
}
//=============================================================
template <class T>
int16_t AudioFile<T>::sampleToSixteenBitInt (T sample)
{
sample = clamp (sample, -1., 1.);
return static_cast<int16_t> (sample * 32767.);
}
//=============================================================
template <class T>
uint8_t AudioFile<T>::sampleToSingleByte (T sample)
{
sample = clamp (sample, -1., 1.);
sample = (sample + 1.) / 2.;
return static_cast<uint8_t> (sample * 255.);
}
//=============================================================
template <class T>
T AudioFile<T>::singleByteToSample (uint8_t sample)
{
return static_cast<T> (sample - 128) / static_cast<T> (128.);
}
//=============================================================
template <class T>
T AudioFile<T>::clamp (T value, T minValue, T maxValue)
{
value = std::min (value, maxValue);
value = std::max (value, minValue);
return value;
}
//=============================================================
template <class T>
void AudioFile<T>::reportError (std::string errorMessage)
{
if (logErrorsToConsole)
std::cout << errorMessage << std::endl;
}
#if defined (_MSC_VER)
__pragma(warning (pop))
#elif defined (__GNUC__)
_Pragma("GCC diagnostic pop")
#endif
#endif /* AudioFile_h */