void HLAudioFile::ReadAudioBytes(SInt64 inOffset, UInt32& ioNumberBytes, void* outData, bool inCache)
{
ThrowIf(mAudioFileID == 0, CAException(fnOpnErr), "HLAudioFile::ReadAudioBytes: file isn't prepared");
OSStatus theError = AudioFileReadBytes(mAudioFileID, inCache, inOffset, &ioNumberBytes, outData);
ThrowIfError(theError, CAException(theError), "HLAudioFile::ReadAudioBytes: couldn't read the data");
}
OSStatus LoadFileData(const char *inFilePath, void* &outData, UInt32 &outDataSize, ALuint &outBufferID)
{
AudioFileID theAFID = 0;
OSStatus result = noErr;
UInt64 theFileSize = 0;
AudioStreamBasicDescription theFileFormat;
result = LoadFileDataInfo(inFilePath, theAFID, theFileFormat, theFileSize);
outDataSize = (UInt32)theFileSize;
AssertNoError("Error loading file info", fail)
outData = malloc(outDataSize);
result = AudioFileReadBytes(theAFID, false, 0, &outDataSize, outData);
AssertNoError("Error reading file data", fail)
if (!TestAudioFormatNativeEndian(theFileFormat) && (theFileFormat.mBitsPerChannel > 8))
return kSoundEngineErrInvalidFileFormat;
alGenBuffers(1, &outBufferID);
AssertNoOALError("Error generating buffer\n", fail);
alBufferDataStaticProc(outBufferID, GetALFormat(theFileFormat), outData, outDataSize, theFileFormat.mSampleRate);
AssertNoOALError("Error attaching data to buffer\n", fail);
AudioFileClose(theAFID);
return result;
fail:
if (theAFID)
AudioFileClose(theAFID);
if (outData)
{
free(outData);
outData = NULL;
}
return result;
}
void* MyGetOpenALAudioData(CFURLRef inFileURL, ALsizei *outDataSize, ALenum *outDataFormat, ALsizei* outSampleRate)
{
OSStatus err = noErr;
UInt64 fileDataSize = 0;
AudioStreamBasicDescription theFileFormat;
UInt32 thePropertySize = sizeof(theFileFormat);
AudioFileID afid = 0;
void* theData = NULL;
// Open a file with ExtAudioFileOpen()
err = AudioFileOpenURL(inFileURL, kAudioFileReadPermission, 0, &afid);
if(err) { printf("MyGetOpenALAudioData: AudioFileOpenURL FAILED, Error = %ld\n", err); goto Exit; }
// Get the audio data format
err = AudioFileGetProperty(afid, kAudioFilePropertyDataFormat, &thePropertySize, &theFileFormat);
if(err) { printf("MyGetOpenALAudioData: AudioFileGetProperty(kAudioFileProperty_DataFormat) FAILED, Error = %ld\n", err); goto Exit; }
if (theFileFormat.mChannelsPerFrame > 2) {
printf("MyGetOpenALAudioData - Unsupported Format, channel count is greater than stereo\n"); goto Exit;
}
if ((theFileFormat.mFormatID != kAudioFormatLinearPCM) || (!TestAudioFormatNativeEndian(theFileFormat))) {
printf("MyGetOpenALAudioData - Unsupported Format, must be little-endian PCM\n"); goto Exit;
}
if ((theFileFormat.mBitsPerChannel != 8) && (theFileFormat.mBitsPerChannel != 16)) {
printf("MyGetOpenALAudioData - Unsupported Format, must be 8 or 16 bit PCM\n"); goto Exit;
}
thePropertySize = sizeof(fileDataSize);
err = AudioFileGetProperty(afid, kAudioFilePropertyAudioDataByteCount, &thePropertySize, &fileDataSize);
if(err) { printf("MyGetOpenALAudioData: AudioFileGetProperty(kAudioFilePropertyAudioDataByteCount) FAILED, Error = %ld\n", err); goto Exit; }
// Read all the data into memory
UInt32 dataSize = fileDataSize;
theData = malloc(dataSize);
if (theData)
{
AudioFileReadBytes(afid, false, 0, &dataSize, theData);
if(err == noErr)
{
// success
*outDataSize = (ALsizei)dataSize;
*outDataFormat = (theFileFormat.mChannelsPerFrame > 1) ? AL_FORMAT_STEREO16 : AL_FORMAT_MONO16;
*outSampleRate = (ALsizei)theFileFormat.mSampleRate;
}
else
{
// failure
free (theData);
theData = NULL; // make sure to return NULL
printf("MyGetOpenALAudioData: ExtAudioFileRead FAILED, Error = %ld\n", err); goto Exit;
}
}
Exit:
// Dispose the ExtAudioFileRef, it is no longer needed
if (afid) AudioFileClose(afid);
return theData;
}
//-----------------------------------------------------------------------------
bool AudioFile::Load()
{
#if MAC
AudioFileID mAudioFileID;
AudioStreamBasicDescription fileDescription, outputFormat;
SInt64 dataSize64;
UInt32 dataSize;
OSStatus err;
UInt32 size;
// ファイルを開く
FSRef ref;
Boolean isDirectory=false;
FSPathMakeRef((const UInt8*)GetFilePath(), &ref, &isDirectory);
err = AudioFileOpen(&ref, fsRdPerm, 0, &mAudioFileID);
if (err) {
//NSLog(@"AudioFileOpen failed");
return false;
}
// 開いたファイルの基本情報を fileDescription へ
size = sizeof(AudioStreamBasicDescription);
err = AudioFileGetProperty(mAudioFileID, kAudioFilePropertyDataFormat,
&size, &fileDescription);
if (err) {
//NSLog(@"AudioFileGetProperty failed");
AudioFileClose(mAudioFileID);
return false;
}
// 開いたファイルのデータ部のバイト数を dataSize へ
size = sizeof(SInt64);
err = AudioFileGetProperty(mAudioFileID, kAudioFilePropertyAudioDataByteCount,
&size, &dataSize64);
if (err) {
//NSLog(@"AudioFileGetProperty failed");
AudioFileClose(mAudioFileID);
return false;
}
dataSize = static_cast<UInt32>(dataSize64);
AudioFileTypeID fileTypeID;
size = sizeof( AudioFileTypeID );
err = AudioFileGetProperty(mAudioFileID, kAudioFilePropertyFileFormat, &size, &fileTypeID);
if (err) {
//NSLog(@"AudioFileGetProperty failed");
AudioFileClose(mAudioFileID);
return false;
}
// Instrument情報を初期化
mInstData.basekey = 60;
mInstData.lowkey = 0;
mInstData.highkey = 127;
mInstData.loop = 0;
//ループポイントの取得
Float64 st_point=0.0,end_point=0.0;
if ( fileTypeID == kAudioFileAIFFType || fileTypeID == kAudioFileAIFCType ) {
//INSTチャンクの取得
AudioFileGetUserDataSize(mAudioFileID, 'INST', 0, &size);
if ( size > 4 ) {
UInt8 *instChunk = new UInt8[size];
AudioFileGetUserData(mAudioFileID, 'INST', 0, &size, instChunk);
//MIDI情報の取得
mInstData.basekey = instChunk[0];
mInstData.lowkey = instChunk[2];
mInstData.highkey = instChunk[3];
if ( instChunk[9] > 0 ) { //ループフラグを確認
//マーカーの取得
UInt32 writable;
err = AudioFileGetPropertyInfo(mAudioFileID, kAudioFilePropertyMarkerList,
&size, &writable);
if (err) {
//NSLog(@"AudioFileGetPropertyInfo failed");
AudioFileClose(mAudioFileID);
return NULL;
}
UInt8 *markersBuffer = new UInt8[size];
AudioFileMarkerList *markers = reinterpret_cast<AudioFileMarkerList*>(markersBuffer);
err = AudioFileGetProperty(mAudioFileID, kAudioFilePropertyMarkerList,
&size, markers);
if (err) {
//NSLog(@"AudioFileGetProperty failed");
AudioFileClose(mAudioFileID);
return NULL;
}
//ループポイントの設定
for (unsigned int i=0; i<markers->mNumberMarkers; i++) {
if (markers->mMarkers[i].mMarkerID == instChunk[11] ) {
st_point = markers->mMarkers[i].mFramePosition;
}
else if (markers->mMarkers[i].mMarkerID == instChunk[13] ) {
end_point = markers->mMarkers[i].mFramePosition;
}
CFRelease(markers->mMarkers[i].mName);
}
if ( st_point < end_point ) {
mInstData.loop = 1;
}
delete [] markersBuffer;
}
delete [] instChunk;
}
}
else if ( fileTypeID == kAudioFileWAVEType ) {
//smplチャンクの取得
AudioFileGetUserDataSize( mAudioFileID, 'smpl', 0, &size );
if ( size >= sizeof(WAV_smpl) ) {
UInt8 *smplChunk = new UInt8[size];
AudioFileGetUserData( mAudioFileID, 'smpl', 0, &size, smplChunk );
WAV_smpl *smpl = (WAV_smpl *)smplChunk;
smpl->loops = EndianU32_LtoN( smpl->loops );
if ( smpl->loops > 0 ) {
mInstData.loop = true;
mInstData.basekey = EndianU32_LtoN( smpl->note );
st_point = EndianU32_LtoN( smpl->start );
end_point = EndianU32_LtoN( smpl->end ) + 1; //SoundForge等では最終ポイントを含める解釈
//end_point = EndianU32_LtoN( smpl->end ); //PeakではなぜかAIFFと同じ
}
else {
mInstData.basekey = EndianU32_LtoN( smpl->note );
}
delete [] smplChunk;
}
}
//容量の制限
SInt64 dataSamples = dataSize / fileDescription.mBytesPerFrame;
if ( dataSamples > MAXIMUM_SAMPLES ) {
dataSize = MAXIMUM_SAMPLES * fileDescription.mBytesPerFrame;
}
if ( st_point > MAXIMUM_SAMPLES ) {
st_point = MAXIMUM_SAMPLES;
}
if ( end_point > MAXIMUM_SAMPLES ) {
end_point = MAXIMUM_SAMPLES;
}
// 波形一時読み込み用メモリを確保
char *fileBuffer;
unsigned int fileBufferSize;
if (mInstData.loop) {
fileBufferSize = dataSize+EXPAND_BUFFER*fileDescription.mBytesPerFrame;
}
else {
fileBufferSize = dataSize;
}
fileBuffer = new char[fileBufferSize];
memset(fileBuffer, 0, fileBufferSize);
// ファイルから波形データの読み込み
err = AudioFileReadBytes(mAudioFileID, false, 0, &dataSize, fileBuffer);
if (err) {
//NSLog(@"AudioFileReadBytes failed");
AudioFileClose(mAudioFileID);
delete [] fileBuffer;
return false;
}
AudioFileClose(mAudioFileID);
//ループを展開する
Float64 adjustment = 1.0;
outputFormat=fileDescription;
if (mInstData.loop) {
UInt32 plusalpha=0, framestocopy;
while (plusalpha < EXPAND_BUFFER) {
framestocopy =
(end_point-st_point)>(EXPAND_BUFFER-plusalpha)?(EXPAND_BUFFER-plusalpha):end_point-st_point;
memcpy(fileBuffer+((int)end_point+plusalpha)*fileDescription.mBytesPerFrame,
fileBuffer+(int)st_point*fileDescription.mBytesPerFrame,
framestocopy*fileDescription.mBytesPerFrame);
plusalpha += framestocopy;
}
dataSize += plusalpha*fileDescription.mBytesPerFrame;
//16サンプル境界にFIXする
adjustment = ( (long long)((end_point-st_point)/16) ) / ((end_point-st_point)/16.0);
st_point *= adjustment;
end_point *= adjustment;
}
outputFormat.mFormatID = kAudioFormatLinearPCM;
outputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagsNativeEndian;
outputFormat.mChannelsPerFrame = 1;
outputFormat.mBytesPerFrame = sizeof(float);
outputFormat.mBitsPerChannel = 32;
outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame;
// バイトオーダー変換用のコンバータを用意
AudioConverterRef converter;
err = AudioConverterNew(&fileDescription, &outputFormat, &converter);
if (err) {
//NSLog(@"AudioConverterNew failed");
delete [] fileBuffer;
return false;
}
//サンプリングレート変換の質を最高に設定
// if (fileDescription.mSampleRate != outputFormat.mSampleRate) {
// size = sizeof(UInt32);
// UInt32 setProp = kAudioConverterQuality_Max;
// AudioConverterSetProperty(converter, kAudioConverterSampleRateConverterQuality,
// size, &setProp);
//
// size = sizeof(UInt32);
// setProp = kAudioConverterSampleRateConverterComplexity_Mastering;
// AudioConverterSetProperty(converter, kAudioConverterSampleRateConverterComplexity,
// size, &setProp);
//
// }
//出力に必要十分なバッファサイズを得る
UInt32 outputSize = dataSize;
size = sizeof(UInt32);
err = AudioConverterGetProperty(converter, kAudioConverterPropertyCalculateOutputBufferSize,
&size, &outputSize);
if (err) {
//NSLog(@"AudioConverterGetProperty failed");
delete [] fileBuffer;
AudioConverterDispose(converter);
return false;
}
UInt32 monoSamples = outputSize/sizeof(float);
// バイトオーダー変換
float *monoData = new float[monoSamples];
AudioConverterConvertBuffer(converter, dataSize, fileBuffer,
&outputSize, monoData);
if(outputSize == 0) {
//NSLog(@"AudioConverterConvertBuffer failed");
delete [] fileBuffer;
AudioConverterDispose(converter);
return false;
}
//ループ長が16の倍数でない場合はサンプリングレート変換
Float64 inputSampleRate = fileDescription.mSampleRate;
Float64 outputSampleRate = fileDescription.mSampleRate * adjustment;
int outSamples = monoSamples;
if ( outputSampleRate == inputSampleRate ) {
m_pAudioData = new short[monoSamples];
for (int i=0; i<monoSamples; i++) {
m_pAudioData[i] = static_cast<short>(monoData[i] * 32768);
}
}
else {
outSamples = static_cast<int>(monoSamples / (inputSampleRate / outputSampleRate));
m_pAudioData = new short[outSamples];
resampling(monoData, monoSamples, inputSampleRate,
m_pAudioData, &outSamples, outputSampleRate);
}
// 後始末
delete [] monoData;
delete [] fileBuffer;
AudioConverterDispose(converter);
//Instデータの設定
if ( st_point > MAXIMUM_SAMPLES ) {
mInstData.lp = MAXIMUM_SAMPLES;
}
else {
mInstData.lp = st_point;
}
if ( end_point > MAXIMUM_SAMPLES ) {
mInstData.lp_end = MAXIMUM_SAMPLES;
}
else {
mInstData.lp_end = end_point;
}
mInstData.srcSamplerate = outputSampleRate;
mLoadedSamples = outSamples;
mIsLoaded = true;
return true;
#else
//Windowsのオーディオファイル読み込み処理
// ファイルを開く
HMMIO hmio = NULL;
MMRESULT err;
DWORD size;
hmio = mmioOpen( mPath, NULL, MMIO_READ );
if ( !hmio ) {
return false;
}
// RIFFチャンクを探す
MMCKINFO riffChunkInfo;
riffChunkInfo.fccType = mmioFOURCC('W', 'A', 'V', 'E');
err = mmioDescend( hmio, &riffChunkInfo, NULL, MMIO_FINDRIFF );
if ( err != MMSYSERR_NOERROR ) {
mmioClose( hmio, 0 );
return false;
}
if ( (riffChunkInfo.ckid != FOURCC_RIFF) || (riffChunkInfo.fccType != mmioFOURCC('W', 'A', 'V', 'E') ) ) {
mmioClose( hmio, 0 );
return false;
}
// フォーマットチャンクを探す
MMCKINFO formatChunkInfo;
formatChunkInfo.ckid = mmioFOURCC('f', 'm', 't', ' ');
err = mmioDescend( hmio, &formatChunkInfo, &riffChunkInfo, MMIO_FINDCHUNK );
if ( err != MMSYSERR_NOERROR ) {
mmioClose( hmio, 0 );
return false;
}
if ( formatChunkInfo.cksize < sizeof(PCMWAVEFORMAT) ) {
mmioClose( hmio, 0 );
return false;
}
//フォーマット情報を取得
WAVEFORMATEX pcmWaveFormat;
DWORD fmsize = (formatChunkInfo.cksize > sizeof(WAVEFORMATEX)) ? sizeof(WAVEFORMATEX):formatChunkInfo.cksize;
size = mmioRead( hmio, (HPSTR)&pcmWaveFormat, fmsize );
if ( size != fmsize ) {
mmioClose( hmio, 0 );
return false;
}
if ( pcmWaveFormat.wFormatTag != WAVE_FORMAT_PCM ) {
mmioClose( hmio, 0 );
return false;
}
mmioAscend(hmio, &formatChunkInfo, 0);
// Instrument情報を初期化
mInstData.basekey = 60;
mInstData.lowkey = 0;
mInstData.highkey = 127;
mInstData.loop = 0;
//smplチャンクを探す
MMCKINFO smplChunkInfo;
smplChunkInfo.ckid = mmioFOURCC('s', 'm', 'p', 'l');
err = mmioDescend( hmio, &smplChunkInfo, &riffChunkInfo, MMIO_FINDCHUNK );
if ( err != MMSYSERR_NOERROR ) {
smplChunkInfo.cksize = 0;
}
double st_point=0.0;
double end_point=0.0;
if ( smplChunkInfo.cksize >= sizeof(WAV_smpl) ) {
//ループポイントの取得
unsigned char *smplChunk = new unsigned char[smplChunkInfo.cksize];
size = mmioRead(hmio,(HPSTR)smplChunk, smplChunkInfo.cksize);
WAV_smpl *smpl = (WAV_smpl *)smplChunk;
if ( smpl->loops > 0 ) {
mInstData.loop = 1;
mInstData.basekey = smpl->note;
st_point = smpl->start;
end_point = smpl->end + 1; //SoundForge等では最終ポイントを含める解釈
}
else {
mInstData.basekey = smpl->note;
}
delete [] smplChunk;
}
mmioAscend(hmio, &formatChunkInfo, 0);
//dataチャンクを探す
MMCKINFO dataChunkInfo;
dataChunkInfo.ckid = mmioFOURCC('d', 'a', 't', 'a');
err = mmioDescend( hmio, &dataChunkInfo, &riffChunkInfo, MMIO_FINDCHUNK );
if( err != MMSYSERR_NOERROR ) {
mmioClose( hmio, 0 );
return false;
}
// 波形一時読み込み用メモリを確保
unsigned int dataSize = dataChunkInfo.cksize;
int bytesPerSample = pcmWaveFormat.nBlockAlign;
char *fileBuffer;
unsigned int fileBufferSize;
//容量制限
int dataSamples = dataSize / pcmWaveFormat.nBlockAlign;
if ( dataSamples > MAXIMUM_SAMPLES ) {
dataSize = MAXIMUM_SAMPLES * pcmWaveFormat.nBlockAlign;
}
if ( st_point > MAXIMUM_SAMPLES ) {
st_point = MAXIMUM_SAMPLES;
}
if ( end_point > MAXIMUM_SAMPLES ) {
end_point = MAXIMUM_SAMPLES;
}
if (mInstData.loop) {
fileBufferSize = dataSize+EXPAND_BUFFER*bytesPerSample;
}
else {
fileBufferSize = dataSize;
}
fileBuffer = new char[fileBufferSize];
memset(fileBuffer, 0, fileBufferSize);
// ファイルから波形データの読み込み
size = mmioRead(hmio, (HPSTR)fileBuffer, dataSize);
if ( size != dataSize ) {
mmioClose( hmio, 0 );
return false;
}
mmioClose(hmio,0);
//ループを展開する
double inputSampleRate = pcmWaveFormat.nSamplesPerSec;
double outputSampleRate = inputSampleRate;
if (mInstData.loop) {
unsigned int plusalpha=0;
double framestocopy;
while (plusalpha < EXPAND_BUFFER) {
framestocopy =
(end_point-st_point)>(EXPAND_BUFFER-plusalpha)?(EXPAND_BUFFER-plusalpha):end_point-st_point;
memcpy(fileBuffer+((int)end_point+plusalpha)*bytesPerSample,
fileBuffer+(int)st_point*bytesPerSample,
static_cast<size_t>(framestocopy*bytesPerSample));
plusalpha += static_cast<unsigned int>(framestocopy);
}
dataSize += plusalpha*bytesPerSample;
//16サンプル境界にFIXする
double adjustment = ( (long long)((end_point-st_point)/16) ) / ((end_point-st_point)/16.0);
outputSampleRate *= adjustment;
st_point *= adjustment;
end_point *= adjustment;
}
//一旦floatモノラルデータに変換
int bytesPerChannel = bytesPerSample / pcmWaveFormat.nChannels;
unsigned int inputPtr = 0;
unsigned int outputPtr = 0;
int monoSamples = dataSize / bytesPerSample;
float range = static_cast<float>((1<<(bytesPerChannel*8-1)) * pcmWaveFormat.nChannels);
float *monoData = new float[monoSamples];
while (inputPtr < dataSize) {
int frameSum = 0;
for (int ch=0; ch<pcmWaveFormat.nChannels; ch++) {
for (int i=0; i<bytesPerChannel; i++) {
if (i<bytesPerChannel-1) {
frameSum += (unsigned char)fileBuffer[inputPtr] << (8*i);
}
else {
frameSum += fileBuffer[inputPtr] << (8*i);
}
inputPtr++;
}
}
monoData[outputPtr] = frameSum / range;
outputPtr++;
}
//ループ長が16の倍数でない場合はサンプリングレート変換
int outSamples = monoSamples;
if ( outputSampleRate == inputSampleRate ) {
m_pAudioData = new short[monoSamples];
for (int i=0; i<monoSamples; i++) {
m_pAudioData[i] = static_cast<short>(monoData[i] * 32768);
}
}
else {
outSamples = static_cast<int>(monoSamples / (inputSampleRate / outputSampleRate));
m_pAudioData = new short[outSamples];
resampling(monoData, monoSamples, inputSampleRate,
m_pAudioData, &outSamples, outputSampleRate);
}
// 後始末
delete [] fileBuffer;
delete [] monoData;
//Instデータの設定
mInstData.lp = static_cast<int>(st_point);
mInstData.lp_end = static_cast<int>(end_point);
mInstData.srcSamplerate = outputSampleRate;
mLoadedSamples = outSamples;
mIsLoaded = true;
return true;
#endif
}
// ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ALUTAPI ALvoid ALUTAPIENTRY alutLoadWAVFile(ALbyte *file,ALenum *format,ALvoid **data,ALsizei *size,ALsizei *freq)
{
OSStatus err = noErr;
AudioFileID audioFile = 0;
FSRef fsRef;
*data = NULL; // in case of failure, do not return some unitialized value as a bogus address
if (IsRelativePath(file))
{
char absolutePath[256];
// we need to make a full path here so FSPathMakeRef() works properly
MakeAbsolutePath(file, absolutePath, 256);
// create an fsref from the file parameter
err = FSPathMakeRef ((const UInt8 *) absolutePath, &fsRef, NULL);
}
else
err = FSPathMakeRef ((const UInt8 *) file, &fsRef, NULL);
if (err == noErr)
{
err = AudioFileOpen(&fsRef, fsRdPerm, 0, &audioFile);
if (err == noErr)
{
UInt32 dataSize;
CAStreamBasicDescription asbd;
dataSize = sizeof(CAStreamBasicDescription);
AudioFileGetProperty(audioFile, kAudioFilePropertyDataFormat, &dataSize, &asbd);
*format = GetOALFormatFromASBD(asbd);
if (IsFormatSupported(*format))
{
*freq = (UInt32) asbd.mSampleRate;
SInt64 audioDataSize = 0;
dataSize = sizeof(audioDataSize);
err = AudioFileGetProperty(audioFile, kAudioFilePropertyAudioDataByteCount, &dataSize, &audioDataSize);
if (err == noErr)
{
*size = audioDataSize;
*data = NULL;
*data = calloc(1, audioDataSize);
if (*data)
{
dataSize = audioDataSize;
err = AudioFileReadBytes(audioFile, false, 0, &dataSize, *data);
if ((asbd.mFormatID == kAudioFormatLinearPCM) && (asbd.mBitsPerChannel > 8))
{
// we just got 16 bit pcm data out of a WAVE file on a big endian platform, so endian swap the data
AudioConverterRef converter;
CAStreamBasicDescription outFormat = asbd;
void * tempData = NULL;
// ste format to big endian
outFormat.mFormatFlags = kAudioFormatFlagIsBigEndian | kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
// make some place for converted data
tempData = calloc(1 , audioDataSize);
err = AudioConverterNew(&asbd, &outFormat, &converter);
if ((err == noErr) && (tempData != NULL))
{
UInt32 bufferSize = audioDataSize;
err = AudioConverterConvertBuffer(converter, audioDataSize, *data, &bufferSize, tempData);
if (err == noErr)
memcpy(*data, tempData, audioDataSize);
AudioConverterDispose(converter);
}
if (tempData) free (tempData);
}
}
}
}
err = AudioFileClose(audioFile);
}
}
}
static bool LoadWAVFile(const char* filename, ALenum* format, ALvoid** data, ALsizei* size, ALsizei* freq, Float64* estimatedDurationOut)
{
CFStringRef filenameStr = CFStringCreateWithCString( NULL, filename, kCFStringEncodingUTF8 );
CFURLRef url = CFURLCreateWithFileSystemPath( NULL, filenameStr, kCFURLPOSIXPathStyle, false );
CFRelease( filenameStr );
AudioFileID audioFile;
OSStatus error = AudioFileOpenURL( url, kAudioFileReadPermission, kAudioFileWAVEType, &audioFile );
CFRelease( url );
if ( error != noErr )
{
fprintf( stderr, "Error opening audio file. %d\n", error );
return false;
}
AudioStreamBasicDescription basicDescription;
UInt32 propertySize = sizeof(basicDescription);
error = AudioFileGetProperty( audioFile, kAudioFilePropertyDataFormat, &propertySize, &basicDescription );
if ( error != noErr )
{
fprintf( stderr, "Error reading audio file basic description. %d\n", error );
AudioFileClose( audioFile );
return false;
}
if ( basicDescription.mFormatID != kAudioFormatLinearPCM )
{
// Need PCM for Open AL. WAVs are (I believe) by definition PCM, so this check isn't necessary. It's just here
// in case I ever use this with another audio format.
fprintf( stderr, "Audio file is not linear-PCM. %d\n", basicDescription.mFormatID );
AudioFileClose( audioFile );
return false;
}
UInt64 audioDataByteCount = 0;
propertySize = sizeof(audioDataByteCount);
error = AudioFileGetProperty( audioFile, kAudioFilePropertyAudioDataByteCount, &propertySize, &audioDataByteCount );
if ( error != noErr )
{
fprintf( stderr, "Error reading audio file byte count. %d\n", error );
AudioFileClose( audioFile );
return false;
}
Float64 estimatedDuration = 0;
propertySize = sizeof(estimatedDuration);
error = AudioFileGetProperty( audioFile, kAudioFilePropertyEstimatedDuration, &propertySize, &estimatedDuration );
if ( error != noErr )
{
fprintf( stderr, "Error reading estimated duration of audio file. %d\n", error );
AudioFileClose( audioFile );
return false;
}
ALenum alFormat = 0;
if ( basicDescription.mChannelsPerFrame == 1 )
{
if ( basicDescription.mBitsPerChannel == 8 )
alFormat = AL_FORMAT_MONO8;
else if ( basicDescription.mBitsPerChannel == 16 )
alFormat = AL_FORMAT_MONO16;
else
{
fprintf( stderr, "Expected 8 or 16 bits for the mono channel but got %d\n", basicDescription.mBitsPerChannel );
AudioFileClose( audioFile );
return false;
}
}
else if ( basicDescription.mChannelsPerFrame == 2 )
{
if ( basicDescription.mBitsPerChannel == 8 )
alFormat = AL_FORMAT_STEREO8;
else if ( basicDescription.mBitsPerChannel == 16 )
alFormat = AL_FORMAT_STEREO16;
else
{
fprintf( stderr, "Expected 8 or 16 bits per channel but got %d\n", basicDescription.mBitsPerChannel );
AudioFileClose( audioFile );
return false;
}
}
else
{
fprintf( stderr, "Expected 1 or 2 channels in audio file but got %d\n", basicDescription.mChannelsPerFrame );
AudioFileClose( audioFile );
return false;
}
UInt32 numBytesToRead = audioDataByteCount;
void* buffer = malloc( numBytesToRead );
if ( buffer == NULL )
{
fprintf( stderr, "Error allocating buffer for audio data of size %u\n", numBytesToRead );
return false;
}
error = AudioFileReadBytes( audioFile, false, 0, &numBytesToRead, buffer );
AudioFileClose( audioFile );
if ( error != noErr )
{
fprintf( stderr, "Error reading audio bytes. %d\n", error );
free(buffer);
return false;
}
if ( numBytesToRead != audioDataByteCount )
{
fprintf( stderr, "Tried to read %lld bytes from the audio file but only got %d bytes\n", audioDataByteCount, numBytesToRead );
free(buffer);
return false;
}
*freq = basicDescription.mSampleRate;
*size = audioDataByteCount;
*format = alFormat;
*data = buffer;
*estimatedDurationOut = estimatedDuration;
return true;
}
