//=======================================================================
/** @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 .
*/
//=======================================================================
#include "AudioFile.h"
#include
#include
#include
//=============================================================
// Pre-defined 10-byte representations of common sample rates
std::unordered_map > 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
AudioFile::AudioFile()
{
bitDepth = 16;
sampleRate = 44100;
samples.resize (1);
samples[0].resize (0);
audioFileFormat = AudioFileFormat::NotLoaded;
}
//=============================================================
template
uint32_t AudioFile::getSampleRate() const
{
return sampleRate;
}
//=============================================================
template
int AudioFile::getNumChannels() const
{
return (int)samples.size();
}
//=============================================================
template
bool AudioFile::isMono() const
{
return getNumChannels() == 1;
}
//=============================================================
template
bool AudioFile::isStereo() const
{
return getNumChannels() == 2;
}
//=============================================================
template
int AudioFile::getBitDepth() const
{
return bitDepth;
}
//=============================================================
template
int AudioFile::getNumSamplesPerChannel() const
{
if (samples.size() > 0)
return (int) samples[0].size();
else
return 0;
}
//=============================================================
template
double AudioFile::getLengthInSeconds() const
{
return (double)getNumSamplesPerChannel() / (double)sampleRate;
}
//=============================================================
template
void AudioFile::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
bool AudioFile::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;
}
int numSamples = (int)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 (int i = 0; i < numSamples; i++)
{
samples[k][i] = newBuffer[k][i];
}
}
return true;
}
//=============================================================
template
void AudioFile::setAudioBufferSize (int numChannels, int numSamples)
{
samples.resize (numChannels);
setNumSamplesPerChannel (numSamples);
}
//=============================================================
template
void AudioFile::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
void AudioFile::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
void AudioFile::setBitDepth (int numBitsPerSample)
{
bitDepth = numBitsPerSample;
}
//=============================================================
template
void AudioFile::setSampleRate (uint32_t newSampleRate)
{
sampleRate = newSampleRate;
}
//=============================================================
template
bool AudioFile::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 begin (file), end;
std::vector 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
bool AudioFile::decodeWaveFile (std::vector& 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);
int16_t 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 = 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
bool AudioFile::decodeAiffFile (std::vector& 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
uint32_t AudioFile::getAiffSampleRate (std::vector& fileData, int sampleRateStartIndex)
{
for (auto it : aiffSampleRateTable)
{
if (tenByteMatch (fileData, sampleRateStartIndex, it.second, 0))
return it.first;
}
return -1;
}
//=============================================================
template
bool AudioFile::tenByteMatch (std::vector& v1, int startIndex1, std::vector& v2, int startIndex2)
{
for (int i = 0; i < 10; i++)
{
if (v1[startIndex1 + i] != v2[startIndex2 + i])
return false;
}
return true;
}
//=============================================================
template
void AudioFile::addSampleRateToAiffData (std::vector& fileData, uint32_t sampleRate)
{
if (aiffSampleRateTable.count (sampleRate) > 0)
{
for (int i = 0; i < 10; i++)
fileData.push_back (aiffSampleRateTable[sampleRate][i]);
}
}
//=============================================================
template
bool AudioFile::save (std::string filePath, AudioFileFormat format)
{
if (format == AudioFileFormat::Wave)
{
return saveToWaveFile (filePath);
}
else if (format == AudioFileFormat::Aiff)
{
return saveToAiffFile (filePath);
}
return false;
}
//=============================================================
template
bool AudioFile::saveToWaveFile (std::string filePath)
{
std::vector 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
bool AudioFile::saveToAiffFile (std::string filePath)
{
std::vector 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
bool AudioFile::writeDataToFile (std::vector& 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
void AudioFile::addStringToFileData (std::vector& fileData, std::string s)
{
for (int i = 0; i < s.length();i++)
fileData.push_back ((uint8_t) s[i]);
}
//=============================================================
template
void AudioFile::addInt32ToFileData (std::vector& 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
void AudioFile::addInt16ToFileData (std::vector& 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
void AudioFile::clearAudioBuffer()
{
for (int i = 0; i < samples.size();i++)
{
samples[i].clear();
}
samples.clear();
}
//=============================================================
template
AudioFileFormat AudioFile::determineAudioFileFormat (std::vector& 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
int32_t AudioFile::fourBytesToInt (std::vector& 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
int16_t AudioFile::twoBytesToInt (std::vector& 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
int AudioFile::getIndexOfString (std::vector& source, std::string stringToSearchFor)
{
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
T AudioFile::sixteenBitIntToSample (int16_t sample)
{
return (T)sample / (T)32768.;
}
//===========================================================
template class AudioFile;
template class AudioFile;