// ____________________________________________________________________________________
// AudioQueue callback function, called when an input buffers has been filled.
static void MyInputBufferHandler( void * inUserData,
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp * inStartTime,
UInt32 inNumPackets,
const AudioStreamPacketDescription *inPacketDesc)
{
MyRecorder *aqr = (MyRecorder *)inUserData;
try {
if (aqr->verbose) {
printf("buf data %p, 0x%x bytes, 0x%x packets\n", inBuffer->mAudioData,
(int)inBuffer->mAudioDataByteSize, (int)inNumPackets);
}
if (inNumPackets > 0) {
// write packets to file
XThrowIfError(AudioFileWritePackets(aqr->recordFile, FALSE, inBuffer->mAudioDataByteSize,
inPacketDesc, aqr->recordPacket, &inNumPackets, inBuffer->mAudioData),
"AudioFileWritePackets failed");
aqr->recordPacket += inNumPackets;
}
// if we're not stopping, re-enqueue the buffe so that it gets filled again
if (aqr->running)
XThrowIfError(AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL), "AudioQueueEnqueueBuffer failed");
}
catch (CAXException e) {
char buf[256];
fprintf(stderr, "MyInputBufferHandler: %s (%s)\n", e.mOperation, e.FormatError(buf));
}
}
void CAAudioFileConverter::OpenOutputFile(const CAStreamBasicDescription &srcFormat, const CAStreamBasicDescription &destFormat, FSRef &destFSRef, CAAudioChannelLayout &destFileLayout)
{
const ConversionParameters ¶ms = mParams;
// output file
if (params.output.filePath == NULL) {
GenerateOutputFileName(params.input.filePath, srcFormat,
destFormat, params.output.fileType, mOutName);
} else
strcpy(mOutName, params.output.filePath);
// deal with pre-existing output file
if (FSPathMakeRef((UInt8 *)mOutName, &destFSRef, NULL) == noErr) {
XThrowIf(!(params.flags & kOpt_OverwriteOutputFile), 1, "overwrite output file");
// not allowed to overwrite
// output file exists - delete it
XThrowIfError(FSDeleteObject(&destFSRef), "delete output file");
}
// get FSRef/CFStringRef for output file
FSRef outFolderFSRef;
CFStringRef outFileName;
XThrowIfError(PosixPathToParentFSRefAndName(mOutName, outFolderFSRef, outFileName), "locate output audio file");
// create the output file
mDestFile.CreateNew(outFolderFSRef, outFileName, params.output.fileType, destFormat, &destFileLayout.Layout());
CFRelease(outFileName);
}
void GetFormatFromInputFile (AudioFileID inputFile, CAStreamBasicDescription & inputFormat)
{
bool doPrint = true;
UInt32 size;
XThrowIfError(AudioFileGetPropertyInfo(inputFile,
kAudioFilePropertyFormatList, &size, NULL), "couldn't get file's format list info");
UInt32 numFormats = size / sizeof(AudioFormatListItem);
AudioFormatListItem *formatList = new AudioFormatListItem [ numFormats ];
XThrowIfError(AudioFileGetProperty(inputFile,
kAudioFilePropertyFormatList, &size, formatList), "couldn't get file's data format");
numFormats = size / sizeof(AudioFormatListItem); // we need to reassess the actual number of formats when we get it
if (numFormats == 1) {
// this is the common case
inputFormat = formatList[0].mASBD;
} else {
if (doPrint) {
printf ("File has a %d layered data format:\n", (int)numFormats);
for (unsigned int i = 0; i < numFormats; ++i)
CAStreamBasicDescription(formatList[i].mASBD).Print();
printf("\n");
}
// now we should look to see which decoders we have on the system
XThrowIfError(AudioFormatGetPropertyInfo(kAudioFormatProperty_DecodeFormatIDs, 0, NULL, &size), "couldn't get decoder id's");
UInt32 numDecoders = size / sizeof(OSType);
OSType *decoderIDs = new OSType [ numDecoders ];
XThrowIfError(AudioFormatGetProperty(kAudioFormatProperty_DecodeFormatIDs, 0, NULL, &size, decoderIDs), "couldn't get decoder id's");
unsigned int i = 0;
for (; i < numFormats; ++i) {
OSType decoderID = formatList[i].mASBD.mFormatID;
bool found = false;
for (unsigned int j = 0; j < numDecoders; ++j) {
if (decoderID == decoderIDs[j]) {
found = true;
break;
}
}
if (found) break;
}
delete [] decoderIDs;
if (i >= numFormats) {
fprintf (stderr, "Cannot play any of the formats in this file\n");
throw kAudioFileUnsupportedDataFormatError;
}
inputFormat = formatList[i].mASBD;
}
delete [] formatList;
}
// _______________________________________________________________________________________
//
// call for existing file, NOT new one - from Open() or Wrap()
void CAAudioFile::GetExistingFileInfo()
{
LOG_FUNCTION("CAAudioFile::GetExistingFileInfo", "%p", this);
UInt32 propertySize;
OSStatus err;
// get mFileDataFormat
propertySize = sizeof(AudioStreamBasicDescription);
XThrowIfError(AudioFileGetProperty(mAudioFile, kAudioFilePropertyDataFormat, &propertySize, &mFileDataFormat), "get audio file's data format");
// get mFileChannelLayout
err = AudioFileGetPropertyInfo(mAudioFile, kAudioFilePropertyChannelLayout, &propertySize, NULL);
if (err == noErr && propertySize > 0) {
AudioChannelLayout *layout = static_cast<AudioChannelLayout *>(malloc(propertySize));
err = AudioFileGetProperty(mAudioFile, kAudioFilePropertyChannelLayout, &propertySize, layout);
if (err == noErr) {
mFileChannelLayout = layout;
#if VERBOSE_CHANNELMAP
printf("existing file's channel layout: %s\n", CAChannelLayouts::ConstantToString(mFileChannelLayout.Tag()));
#endif
}
free(layout);
XThrowIfError(err, "get audio file's channel layout");
}
if (mMode != kReading)
return;
#if 0
// get mNumberPackets
propertySize = sizeof(mNumberPackets);
XThrowIfError(AudioFileGetProperty(mAudioFile, kAudioFilePropertyAudioDataPacketCount, &propertySize, &mNumberPackets), "get audio file's packet count");
#if VERBOSE_IO
printf("CAAudioFile::GetExistingFileInfo: %qd packets\n", mNumberPackets);
#endif
#endif
// get mMagicCookie
err = AudioFileGetPropertyInfo(mAudioFile, kAudioFilePropertyMagicCookieData, &propertySize, NULL);
if (err == noErr && propertySize > 0) {
mMagicCookie = new Byte[propertySize];
mMagicCookieSize = propertySize;
XThrowIfError(AudioFileGetProperty(mAudioFile, kAudioFilePropertyMagicCookieData, &propertySize, mMagicCookie), "get audio file's magic cookie");
}
InitFileMaxPacketSize();
mPacketMark = 0;
mFrameMark = 0;
UpdateClientMaxPacketSize();
}
// _______________________________________________________________________________________
//
CFArrayRef CAAudioFile::GetConverterConfig()
{
CFArrayRef plist;
UInt32 propertySize = sizeof(plist);
XThrowIfError(AudioConverterGetProperty(mConverter, kAudioConverterPropertySettings, &propertySize, &plist), "get converter property settings");
return plist;
}
// _______________________________________________________________________________________
//
void CAAudioFile::FlushEncoder()
{
if (mConverter != NULL) {
mFinishingEncoding = true;
WritePacketsFromCallback(WriteInputProc, this);
mFinishingEncoding = false;
// get priming info from converter, set it on the file
if (mFileDataFormat.mBitsPerChannel == 0) {
UInt32 propertySize;
OSStatus err;
AudioConverterPrimeInfo primeInfo;
propertySize = sizeof(primeInfo);
err = AudioConverterGetProperty(mConverter, kAudioConverterPrimeInfo, &propertySize, &primeInfo);
if (err == noErr) {
AudioFilePacketTableInfo pti;
propertySize = sizeof(pti);
err = AudioFileGetProperty(mAudioFile, kAudioFilePropertyPacketTableInfo, &propertySize, &pti);
if (err == noErr) {
//printf("old packet table info: %qd valid, %ld priming, %ld remainder\n", pti.mNumberValidFrames, pti.mPrimingFrames, pti.mRemainderFrames);
UInt64 totalFrames = pti.mNumberValidFrames + pti.mPrimingFrames + pti.mRemainderFrames;
pti.mPrimingFrames = primeInfo.leadingFrames;
pti.mRemainderFrames = primeInfo.trailingFrames;
pti.mNumberValidFrames = totalFrames - pti.mPrimingFrames - pti.mRemainderFrames;
//printf("new packet table info: %qd valid, %ld priming, %ld remainder\n", pti.mNumberValidFrames, pti.mPrimingFrames, pti.mRemainderFrames);
XThrowIfError(AudioFileSetProperty(mAudioFile, kAudioFilePropertyPacketTableInfo, sizeof(pti), &pti), "couldn't set packet table info on audio file");
}
}
}
}
}
// _______________________________________________________________________________________
//
void CAAudioFile::Write(UInt32 numPackets, const AudioBufferList *data)
{
if (mIOBufferList.mBuffers[0].mData == NULL) {
#if DEBUG
printf("warning: CAAudioFile::AllocateBuffers called from WritePackets\n");
#endif
AllocateBuffers();
}
if (mMode == kPreparingToWrite)
mMode = kWriting;
else
XThrowIf(mMode != kWriting, kExtAudioFileError_InvalidOperationOrder, "can't write to this file");
if (mConverter != NULL) {
mWritePackets = numPackets;
mWriteBufferList->SetFrom(data);
WritePacketsFromCallback(WriteInputProc, this);
} else {
StartTiming(this, write);
XThrowIfError(AudioFileWritePackets(mAudioFile, mUseCache, data->mBuffers[0].mDataByteSize,
NULL, mPacketMark, &numPackets, data->mBuffers[0].mData),
"write audio file");
ElapsedTime(this, write, mTicksInIO);
#if VERBOSE_IO
printf("CAAudioFile::WritePackets: wrote %ld packets at %qd, %ld bytes\n", numPackets, mPacketMark, data->mBuffers[0].mDataByteSize);
#endif
//mNumberPackets =
mPacketMark += numPackets;
if (mFileDataFormat.mFramesPerPacket > 0)
mFrameMark += numPackets * mFileDataFormat.mFramesPerPacket;
// else: shouldn't happen since we're only called when there's no converter
}
}
// _______________________________________________________________________________________
//
void CAAudioFile::UpdateClientMaxPacketSize()
{
LOG_FUNCTION("CAAudioFile::UpdateClientMaxPacketSize", "%p", this);
mFrame0Offset = 0;
if (mConverter != NULL) {
AudioConverterPropertyID property = (mMode == kReading) ?
kAudioConverterPropertyMaximumOutputPacketSize :
kAudioConverterPropertyMaximumInputPacketSize;
UInt32 propertySize = sizeof(UInt32);
XThrowIfError(AudioConverterGetProperty(mConverter, property, &propertySize, &mClientMaxPacketSize),
"get audio converter's maximum packet size");
if (mFileDataFormat.mBitsPerChannel == 0) {
AudioConverterPrimeInfo primeInfo;
propertySize = sizeof(primeInfo);
OSStatus err = AudioConverterGetProperty(mConverter, kAudioConverterPrimeInfo, &propertySize, &primeInfo);
if (err == noErr)
mFrame0Offset = primeInfo.leadingFrames;
#if VERBOSE_CONVERTER
printf("kAudioConverterPrimeInfo: err = %ld, leadingFrames = %ld\n", err, mFrame0Offset);
#endif
}
} else {
mClientMaxPacketSize = mFileMaxPacketSize;
}
}
// ____________________________________________________________________________________
// Copy a queue's encoder's magic cookie to an audio file.
static void MyCopyEncoderCookieToFile(AudioQueueRef theQueue, AudioFileID theFile)
{
OSStatus err;
UInt32 propertySize;
// get the magic cookie, if any, from the converter
err = AudioQueueGetPropertySize(theQueue, kAudioConverterCompressionMagicCookie, &propertySize);
if (err == noErr && propertySize > 0) {
// there is valid cookie data to be fetched; get it
Byte *magicCookie = (Byte *)malloc(propertySize);
try {
XThrowIfError(AudioQueueGetProperty(theQueue, kAudioConverterCompressionMagicCookie, magicCookie,
&propertySize), "get audio converter's magic cookie");
// now set the magic cookie on the output file
// even though some formats have cookies, some files don't take them, so we ignore the error
/*err =*/ AudioFileSetProperty(theFile, kAudioFilePropertyMagicCookieData, propertySize, magicCookie);
}
catch (CAXException e) {
char buf[256];
fprintf(stderr, "MyCopyEncoderCookieToFile: %s (%s)\n", e.mOperation, e.FormatError(buf));
}
catch (...) {
fprintf(stderr, "MyCopyEncoderCookieToFile: Unexpected exception\n");
}
free(magicCookie);
}
}
// ____________________________________________________________________________________
// Determine the size, in bytes, of a buffer necessary to represent the supplied number
// of seconds of audio data.
static int MyComputeRecordBufferSize(const AudioStreamBasicDescription *format, AudioQueueRef queue, float seconds)
{
int packets, frames, bytes;
frames = (int)ceil(seconds * format->mSampleRate);
if (format->mBytesPerFrame > 0)
bytes = frames * format->mBytesPerFrame;
else {
UInt32 maxPacketSize;
if (format->mBytesPerPacket > 0)
maxPacketSize = format->mBytesPerPacket; // constant packet size
else {
UInt32 propertySize = sizeof(maxPacketSize);
XThrowIfError(AudioQueueGetProperty(queue, kAudioConverterPropertyMaximumOutputPacketSize, &maxPacketSize,
&propertySize), "couldn't get queue's maximum output packet size");
}
if (format->mFramesPerPacket > 0)
packets = frames / format->mFramesPerPacket;
else
packets = frames; // worst-case scenario: 1 frame in a packet
if (packets == 0) // sanity check
packets = 1;
bytes = packets * maxPacketSize;
}
return bytes;
}
void CAAudioFileConverter::ReadCAFInfo()
{
FSRef fsref;
AudioFileID afid = 0;
CAFSourceInfo info;
UInt32 size;
OSStatus err;
try {
XThrowIfError(FSPathMakeRef((UInt8 *)mParams.input.filePath, &fsref, NULL), "couldn't locate input file");
XThrowIfError(AudioFileOpen(&fsref, fsRdPerm, 0, &afid), "couldn't open input file");
size = sizeof(AudioFileTypeID);
XThrowIfError(AudioFileGetProperty(afid, kAudioFilePropertyFileFormat, &size, &info.filetype), "couldn't get input file's format");
if (info.filetype == kAudioFileCAFType) {
size = sizeof(info);
err = AudioFileGetUserData(afid, 'srcI', 0, &size, &info);
if (!err) {
// restore the following from the original file info:
// filetype
// data format
// filename
AudioStreamBasicDescription destfmt;
ASBD_BtoN((AudioStreamBasicDescription *)info.asbd, &destfmt);
mParams.output.dataFormat = destfmt;
mParams.output.fileType = EndianU32_BtoN(info.filetype);
if (mParams.output.filePath == NULL) {
int len = strlen(mParams.input.filePath) + strlen(info.filename) + 2;
char *newname = (char *)malloc(len); // $$$ leaked
const char *dir = dirname(mParams.input.filePath);
if (dir && (dir[0] !='.' && dir[1] != '/'))
sprintf(newname, "%s/%s", dir, info.filename);
else
strcpy(newname, info.filename);
mParams.output.filePath = newname;
mParams.flags = (mParams.flags & ~kOpt_OverwriteOutputFile) | kOpt_NoSanitizeOutputFormat;
}
}
}
AudioFileClose(afid);
}
catch (...) {
if (afid)
AudioFileClose(afid);
throw;
}
}
// _______________________________________________________________________________________
//
SInt64 CAAudioFile::FileDataOffset()
{
if (mFileDataOffset < 0) {
UInt32 propertySize = sizeof(SInt64);
XThrowIfError(AudioFileGetProperty(mAudioFile, kAudioFilePropertyDataOffset, &propertySize, &mFileDataOffset), "couldn't get file's data offset");
}
return mFileDataOffset;
}
int main (int argc, char * const argv[])
{
char inputFile[]="blip.mp3";
static const double threshold=0.50;
int hardware=macbookpro;
double x,y,z,prev_x,prev_y,prev_z;
AudioFileID audioFile;
CFURLRef theURL = CFURLCreateFromFileSystemRepresentation(kCFAllocatorDefault, (UInt8*)inputFile, strlen(inputFile), false);
XThrowIfError (AudioFileOpenURL (theURL, kAudioFileReadPermission, 0, &audioFile), "AudioFileOpenURL");
// get the number of channels of the file
CAStreamBasicDescription fileFormat;
UInt32 propsize = sizeof(CAStreamBasicDescription);
XThrowIfError (AudioFileGetProperty(audioFile, kAudioFilePropertyDataFormat, &propsize, &fileFormat), "AudioFileGetProperty");
// lets set up our playing state now
AUGraph theGraph;
CAAudioUnit fileAU;
// this makes the graph, the file AU and sets it all up for playing
MakeSimpleGraph (theGraph, fileAU, fileFormat, audioFile);
// now we load the file contents up for playback before we start playing
// this has to be done the AU is initialized and anytime it is reset or uninitialized
Float64 fileDuration = PrepareFileAU (fileAU, fileFormat, audioFile);
printf ("file duration: %f secs\n", fileDuration);
read_sms_real(hardware,&x,&y,&z);
prev_x=x;
prev_y=y;
prev_z=z;
for(;;) {
read_sms_real(hardware,&x,&y,&z);
//printf("x: %f y: %f z: %f\n",x,y,z);
if(isDelta(threshold,x,y,z,prev_x,prev_y,prev_z))
XThrowIfError (AUGraphStart (theGraph), "AUGraphStart");
prev_x=x;
prev_y=y;
prev_z=z;
}
// sleep until the file is finished
//usleep ((int)(fileDuration * 1000. * 1000.));
// lets clean up
XThrowIfError (AUGraphStop (theGraph), "AUGraphStop");
XThrowIfError (AUGraphUninitialize (theGraph), "AUGraphUninitialize");
XThrowIfError (AudioFileClose (audioFile), "AudioFileClose");
XThrowIfError (AUGraphClose (theGraph), "AUGraphClose");
return 0;
}
// _______________________________________________________________________________________
//
void CAAudioFile::Open(const FSRef &fsref)
{
LOG_FUNCTION("CAAudioFile::Open", "%p", this);
XThrowIf(mMode != kClosed, kExtAudioFileError_InvalidOperationOrder, "file already open");
mFSRef = fsref;
XThrowIfError(AudioFileOpen(&mFSRef, fsRdPerm, 0, &mAudioFile), "open audio file");
mOwnOpenFile = true;
mMode = kReading;
GetExistingFileInfo();
}
void CAAudioFileConverter::WriteCAFInfo()
{
FSRef fsref;
AudioFileID afid = 0;
CAFSourceInfo info;
UInt32 size;
try {
XThrowIfError(FSPathMakeRef((UInt8 *)mParams.input.filePath, &fsref, NULL), "couldn't locate input file");
XThrowIfError(AudioFileOpen(&fsref, fsRdPerm, 0, &afid), "couldn't open input file");
size = sizeof(AudioFileTypeID);
XThrowIfError(AudioFileGetProperty(afid, kAudioFilePropertyFileFormat, &size, &info.filetype), "couldn't get input file's format");
AudioFileClose(afid);
afid = 0;
XThrowIfError(FSPathMakeRef((UInt8 *)mOutName, &fsref, NULL), "couldn't locate output file");
XThrowIfError(AudioFileOpen(&fsref, fsRdWrPerm, 0, &afid), "couldn't open output file");
const char *srcFilename = strrchr(mParams.input.filePath, '/');
if (srcFilename++ == NULL) srcFilename = mParams.input.filePath;
ASBD_NtoB(&mSrcFormat, (AudioStreamBasicDescription *)info.asbd);
int namelen = std::min(kMaxFilename-1, (int)strlen(srcFilename));
memcpy(info.filename, srcFilename, namelen);
info.filename[namelen++] = 0;
info.filetype = EndianU32_NtoB(info.filetype);
XThrowIfError(AudioFileSetUserData(afid, 'srcI', 0, offsetof(CAFSourceInfo, filename) + namelen, &info), "couldn't set CAF file's source info chunk");
AudioFileClose(afid);
}
catch (...) {
if (afid)
AudioFileClose(afid);
throw;
}
}
double PrepareFileAU (CAAudioUnit &au, CAStreamBasicDescription &fileFormat, AudioFileID audioFile)
{
//
// calculate the duration
UInt64 nPackets;
UInt32 propsize = sizeof(nPackets);
XThrowIfError (AudioFileGetProperty(audioFile, kAudioFilePropertyAudioDataPacketCount, &propsize, &nPackets), "kAudioFilePropertyAudioDataPacketCount");
Float64 fileDuration = (nPackets * fileFormat.mFramesPerPacket) / fileFormat.mSampleRate;
ScheduledAudioFileRegion rgn;
memset (&rgn.mTimeStamp, 0, sizeof(rgn.mTimeStamp));
rgn.mTimeStamp.mFlags = kAudioTimeStampSampleTimeValid;
rgn.mTimeStamp.mSampleTime = 0;
rgn.mCompletionProc = NULL;
rgn.mCompletionProcUserData = NULL;
rgn.mAudioFile = audioFile;
rgn.mLoopCount = 1;
rgn.mStartFrame = 0;
rgn.mFramesToPlay = UInt32(nPackets * fileFormat.mFramesPerPacket);
// tell the file player AU to play all of the file
XThrowIfError (au.SetProperty (kAudioUnitProperty_ScheduledFileRegion,
kAudioUnitScope_Global, 0,&rgn, sizeof(rgn)), "kAudioUnitProperty_ScheduledFileRegion");
// prime the fp AU with default values
UInt32 defaultVal = 0;
XThrowIfError (au.SetProperty (kAudioUnitProperty_ScheduledFilePrime,
kAudioUnitScope_Global, 0, &defaultVal, sizeof(defaultVal)), "kAudioUnitProperty_ScheduledFilePrime");
// tell the fp AU when to start playing (this ts is in the AU's render time stamps; -1 means next render cycle)
AudioTimeStamp startTime;
memset (&startTime, 0, sizeof(startTime));
startTime.mFlags = kAudioTimeStampSampleTimeValid;
startTime.mSampleTime = -1;
XThrowIfError (au.SetProperty(kAudioUnitProperty_ScheduleStartTimeStamp,
kAudioUnitScope_Global, 0, &startTime, sizeof(startTime)), "kAudioUnitProperty_ScheduleStartTimeStamp");
return fileDuration;
}
// _______________________________________________________________________________________
//
void CAAudioFile::InitFileMaxPacketSize()
{
LOG_FUNCTION("CAAudioFile::InitFileMaxPacketSize", "%p", this);
UInt32 propertySize = sizeof(UInt32);
OSStatus err = AudioFileGetProperty(mAudioFile, kAudioFilePropertyMaximumPacketSize,
&propertySize, &mFileMaxPacketSize);
if (err) {
// workaround for 3361377: not all file formats' maximum packet sizes are supported
if (!mFileDataFormat.IsPCM())
XThrowIfError(err, "get audio file's maximum packet size");
mFileMaxPacketSize = mFileDataFormat.mBytesPerFrame;
}
AllocateBuffers(true /* okToFail */);
}
// _______________________________________________________________________________________
//
void CAAudioFile::WritePacketsFromCallback(
AudioConverterComplexInputDataProc inInputDataProc,
void * inInputDataProcUserData)
{
while (true) {
// keep writing until we exhaust the input (temporary stop), or produce no output (EOF)
UInt32 numEncodedPackets = mIOBufferSizePackets;
mIOBufferList.mBuffers[0].mDataByteSize = mIOBufferSizeBytes;
#if CAAUDIOFILE_PROFILE
mInConverter = true;
#endif
StartTiming(this, fill);
OSStatus err = AudioConverterFillComplexBuffer(mConverter, inInputDataProc, inInputDataProcUserData,
&numEncodedPackets, &mIOBufferList, mPacketDescs);
ElapsedTime(this, fill, mTicksInConverter);
#if CAAUDIOFILE_PROFILE
mInConverter = false;
#endif
XThrowIf(err != 0 && err != kNoMoreInputRightNow, err, "convert audio packets (write)");
if (numEncodedPackets == 0)
break;
Byte *buf = (Byte *)mIOBufferList.mBuffers[0].mData;
#if VERBOSE_IO
printf("CAAudioFile::WritePacketsFromCallback: wrote %ld packets, %ld bytes\n", numEncodedPackets, mIOBufferList.mBuffers[0].mDataByteSize);
if (mPacketDescs) {
for (UInt32 i = 0; i < numEncodedPackets; ++i) {
printf(" write packet %qd : offset %qd, length %ld\n", mPacketMark + i, mPacketDescs[i].mStartOffset, mPacketDescs[i].mDataByteSize);
#if VERBOSE_IO >= 2
hexdump(buf + mPacketDescs[i].mStartOffset, mPacketDescs[i].mDataByteSize);
#endif
}
}
#endif
StartTiming(this, write);
XThrowIfError(AudioFileWritePackets(mAudioFile, mUseCache, mIOBufferList.mBuffers[0].mDataByteSize, mPacketDescs, mPacketMark, &numEncodedPackets, buf), "write audio file");
ElapsedTime(this, write, mTicksInIO);
mPacketMark += numEncodedPackets;
//mNumberPackets += numEncodedPackets;
if (mFileDataFormat.mFramesPerPacket > 0)
mFrameMark += numEncodedPackets * mFileDataFormat.mFramesPerPacket;
else {
for (UInt32 i = 0; i < numEncodedPackets; ++i)
mFrameMark += mPacketDescs[i].mVariableFramesInPacket;
}
if (err == kNoMoreInputRightNow)
break;
}
}
int SetupRemoteIO (AudioUnit& inRemoteIOUnit, Float64 sampleRate, AURenderCallbackStruct inRenderProc, CAStreamBasicDescription& outFormat)
{
try {
// Open the output unit
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
AudioComponent comp = AudioComponentFindNext(NULL, &desc);
XThrowIfError(AudioComponentInstanceNew(comp, &inRemoteIOUnit), "couldn't open the remote I/O unit");
UInt32 zero = 0;
UInt32 one = 1;
//enable input
XThrowIfError(AudioUnitSetProperty(inRemoteIOUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &one, sizeof(one)), "couldn't enable input on the remote I/O unit");
//disable output
XThrowIfError(AudioUnitSetProperty(inRemoteIOUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &zero, sizeof(zero)), "couldn't disable output ");
//set input callback
XThrowIfError(AudioUnitSetProperty(inRemoteIOUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 1, &inRenderProc, sizeof(inRenderProc)), "couldn't set remote i/o input callback");
// set our required format - LPCM non-interleaved 32 bit floating point
AudioStreamBasicDescription outFormat;
outFormat.mSampleRate = sampleRate;
outFormat.mFormatID = kAudioFormatLinearPCM;
outFormat.mFormatFlags = kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked | kAudioFormatFlagIsFloat | kAudioFormatFlagIsNonInterleaved;
outFormat.mFramesPerPacket = 1;
outFormat.mBytesPerPacket= 4;
outFormat.mChannelsPerFrame = 1;
outFormat.mBitsPerChannel = 32;
outFormat.mBytesPerFrame = 4;
XThrowIfError(AudioUnitSetProperty(inRemoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &outFormat, sizeof(outFormat)), "couldn't set the remote I/O unit's output client format");
XThrowIfError(AudioUnitInitialize(inRemoteIOUnit), "couldn't initialize the remote I/O unit");
}
catch (CAXException &e) {
char buf[256];
fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
return 1;
}
catch (...) {
fprintf(stderr, "An unknown error occurred\n");
return 1;
}
return 0;
}
SInt64 CAAudioFile::FrameToPacket(SInt64 inFrame) const
{
AudioFramePacketTranslation trans;
UInt32 propertySize;
switch (mFileDataFormat.mFramesPerPacket) {
case 1:
return inFrame;
case 0:
trans.mFrame = inFrame;
propertySize = sizeof(trans);
XThrowIfError(AudioFileGetProperty(mAudioFile, kAudioFilePropertyFrameToPacket, &propertySize, &trans),
"packet <-> frame translation unimplemented for format with variable frames/packet");
return trans.mPacket;
}
return inFrame / mFileDataFormat.mFramesPerPacket;
}
// _______________________________________________________________________________________
//
OSStatus CAAudioFile::SetConverterProperty(
AudioConverterPropertyID inPropertyID,
UInt32 inPropertyDataSize,
const void* inPropertyData,
bool inCanFail)
{
OSStatus err = noErr;
//LOG_FUNCTION("ExtAudioFile::SetConverterProperty", "%p %-4.4s", this, (char *)&inPropertyID);
if (inPropertyID == kAudioConverterPropertySettings && *(CFPropertyListRef *)inPropertyData == NULL)
;
else {
err = AudioConverterSetProperty(mConverter, inPropertyID, inPropertyDataSize, inPropertyData);
if (!inCanFail) {
XThrowIfError(err, "set audio converter property");
}
}
UpdateClientMaxPacketSize();
if (mMode == kPreparingToWrite)
FileFormatChanged();
return err;
}
// _______________________________________________________________________________________
// Allocates: mIOBufferList, mIOBufferSizePackets, mPacketDescs
// Dependent on: mFileMaxPacketSize, mIOBufferSizeBytes
void CAAudioFile::AllocateBuffers(bool okToFail)
{
LOG_FUNCTION("CAAudioFile::AllocateBuffers", "%p", this);
if (mFileMaxPacketSize == 0) {
if (okToFail)
return;
XThrowIf(true, kExtAudioFileError_MaxPacketSizeUnknown, "file's maximum packet size is 0");
}
UInt32 bufferSizeBytes = mIOBufferSizeBytes = std::max(mIOBufferSizeBytes, mFileMaxPacketSize);
// must be big enough for at least one maximum size packet
if (mIOBufferList.mBuffers[0].mDataByteSize != bufferSizeBytes) {
mIOBufferList.mNumberBuffers = 1;
mIOBufferList.mBuffers[0].mNumberChannels = mFileDataFormat.mChannelsPerFrame;
if (!mClientOwnsIOBuffer) {
//printf("reallocating I/O buffer\n");
delete[] (Byte *)mIOBufferList.mBuffers[0].mData;
mIOBufferList.mBuffers[0].mData = new Byte[bufferSizeBytes];
}
mIOBufferList.mBuffers[0].mDataByteSize = bufferSizeBytes;
mIOBufferSizePackets = bufferSizeBytes / mFileMaxPacketSize;
}
UInt32 propertySize = sizeof(UInt32);
UInt32 externallyFramed;
XThrowIfError(AudioFormatGetProperty(kAudioFormatProperty_FormatIsExternallyFramed,
sizeof(AudioStreamBasicDescription), &mFileDataFormat, &propertySize, &externallyFramed),
"is format externally framed");
if (mNumPacketDescs != (externallyFramed ? mIOBufferSizePackets : 0)) {
delete[] mPacketDescs;
mPacketDescs = NULL;
mNumPacketDescs = 0;
if (externallyFramed) {
//printf("reallocating packet descs\n");
mPacketDescs = new AudioStreamPacketDescription[mIOBufferSizePackets];
mNumPacketDescs = mIOBufferSizePackets;
}
}
}
void CAAudioFileRecorder::SetFile(const FSRef &parentFSRef, CFStringRef filename, AudioFileTypeID filetype, const CAStreamBasicDescription &dataFormat, const CAAudioChannelLayout *layout)
{
delete mAudioInputPtrs; mAudioInputPtrs = NULL;
CAStreamBasicDescription fileDataFormat = dataFormat;
if (fileDataFormat.mSampleRate == 0.)
fileDataFormat.mSampleRate = mInputDataFormat.mSampleRate;
CAAudioFileWriter::SetFile(parentFSRef, filename, filetype, fileDataFormat, layout);
const CAStreamBasicDescription &fmt = GetFile().GetClientDataFormat();
XThrowIfError(AudioUnitSetProperty(
mInputUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
1,
(void *)&fmt,
sizeof(AudioStreamBasicDescription)), "set audio input format");
GetFile().SetIOBufferSizeBytes(GetBufferSizeFrames() * fmt.mBytesPerFrame);
mAudioInputPtrs = CABufferList::New("audio input ptrs", fmt);
}
int SetupRemoteIO (AudioUnit& inRemoteIOUnit, AURenderCallbackStruct inRenderProc, CAStreamBasicDescription& outFormat)
{
try {
// Open the output unit
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
AudioComponent comp = AudioComponentFindNext(NULL, &desc);
XThrowIfError(AudioComponentInstanceNew(comp, &inRemoteIOUnit), "couldn't open the remote I/O unit");
UInt32 one = 1;
XThrowIfError(AudioUnitSetProperty(inRemoteIOUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &one, sizeof(one)), "couldn't enable input on the remote I/O unit");
XThrowIfError(AudioUnitSetProperty(inRemoteIOUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &inRenderProc, sizeof(inRenderProc)), "couldn't set remote i/o render callback");
// NEWL: Establecer el formato canónico no de los AudioUnits sino del sistema de entrada/salida:
// LPCM, no entrelazado, datos enteros con signo de 16 bits.
// outFormat.SetCanonical(2, false);
// OLDL: set our required format - Canonical AU format: LPCM non-interleaved 8.24 fixed point
outFormat.SetAUCanonical(2, false);
outFormat.mSampleRate = SAMPRATE;
XThrowIfError(AudioUnitSetProperty(inRemoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &outFormat, sizeof(outFormat)), "couldn't set the remote I/O unit's output client format");
XThrowIfError(AudioUnitSetProperty(inRemoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &outFormat, sizeof(outFormat)), "couldn't set the remote I/O unit's input client format");
XThrowIfError(AudioUnitInitialize(inRemoteIOUnit), "couldn't initialize the remote I/O unit");
}
catch (CAXException &e) {
char buf[256];
fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
return 1;
}
catch (...) {
fprintf(stderr, "An unknown error occurred\n");
return 1;
}
return 0;
}
OSStatus DoConvertFile(CFURLRef sourceURL, CFURLRef destinationURL, OSType outputFormat, Float64 outputSampleRate, UInt32 outputBitRate)
{
AudioFileID sourceFileID = 0;
AudioFileID destinationFileID = 0;
AudioConverterRef converter = NULL;
Boolean canResumeFromInterruption = true; // we can continue unless told otherwise
CAStreamBasicDescription srcFormat, dstFormat;
AudioFileIO afio = {};
char *outputBuffer = NULL;
AudioStreamPacketDescription *outputPacketDescriptions = NULL;
OSStatus error = noErr;
// in this sample we should never be on the main thread here
assert(![NSThread isMainThread]);
// transition thread state to kStateRunning before continuing
printf("\nDoConvertFile\n");
try {
// get the source file
XThrowIfError(AudioFileOpenURL(sourceURL, kAudioFileReadPermission, 0, &sourceFileID), "AudioFileOpenURL failed");
// get the source data format
UInt32 size = sizeof(srcFormat);
XThrowIfError(AudioFileGetProperty(sourceFileID, kAudioFilePropertyDataFormat, &size, &srcFormat), "couldn't get source data format");
// setup the output file format
dstFormat.mSampleRate = (outputSampleRate == 0 ? srcFormat.mSampleRate : outputSampleRate); // set sample rate
if (outputFormat == kAudioFormatLinearPCM) {
// if the output format is PC create a 16-bit int PCM file format description as an example
dstFormat.mFormatID = outputFormat;
dstFormat.mChannelsPerFrame = srcFormat.NumberChannels();
dstFormat.mBitsPerChannel = 16;
dstFormat.mBytesPerPacket = dstFormat.mBytesPerFrame = 2 * dstFormat.mChannelsPerFrame;
dstFormat.mFramesPerPacket = 1;
dstFormat.mFormatFlags = kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger; // little-endian
} else {
// compressed format - need to set at least format, sample rate and channel fields for kAudioFormatProperty_FormatInfo
dstFormat.mFormatID = outputFormat;
dstFormat.mChannelsPerFrame = (outputFormat == kAudioFormatiLBC ? 1 : srcFormat.NumberChannels()); // for iLBC num channels must be 1
// use AudioFormat API to fill out the rest of the description
size = sizeof(dstFormat);
XThrowIfError(AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &dstFormat), "couldn't create destination data format");
}
printf("Source File format: "); srcFormat.Print();
printf("Destination format: "); dstFormat.Print();
// create the AudioConverter
XThrowIfError(AudioConverterNew(&srcFormat, &dstFormat, &converter), "AudioConverterNew failed!");
// if the source has a cookie, get it and set it on the Audio Converter
ReadCookie(sourceFileID, converter);
// get the actual formats back from the Audio Converter
size = sizeof(srcFormat);
XThrowIfError(AudioConverterGetProperty(converter, kAudioConverterCurrentInputStreamDescription, &size, &srcFormat), "AudioConverterGetProperty kAudioConverterCurrentInputStreamDescription failed!");
size = sizeof(dstFormat);
XThrowIfError(AudioConverterGetProperty(converter, kAudioConverterCurrentOutputStreamDescription, &size, &dstFormat), "AudioConverterGetProperty kAudioConverterCurrentOutputStreamDescription failed!");
printf("Formats returned from AudioConverter:\n");
printf(" Source format: "); srcFormat.Print();
printf(" Destination File format: "); dstFormat.Print();
// if encoding to AAC set the bitrate
// kAudioConverterEncodeBitRate is a UInt32 value containing the number of bits per second to aim for when encoding data
// when you explicitly set the bit rate and the sample rate, this tells the encoder to stick with both bit rate and sample rate
// but there are combinations (also depending on the number of channels) which will not be allowed
// if you do not explicitly set a bit rate the encoder will pick the correct value for you depending on samplerate and number of channels
// bit rate also scales with the number of channels, therefore one bit rate per sample rate can be used for mono cases
// and if you have stereo or more, you can multiply that number by the number of channels.
if (outputBitRate == 0) {
outputBitRate = 192000; // 192kbs
}
if (dstFormat.mFormatID == kAudioFormatMPEG4AAC) {
UInt32 propSize = sizeof(outputBitRate);
// set the bit rate depending on the samplerate chosen
XThrowIfError(AudioConverterSetProperty(converter, kAudioConverterEncodeBitRate, propSize, &outputBitRate),
"AudioConverterSetProperty kAudioConverterEncodeBitRate failed!");
// get it back and print it out
AudioConverterGetProperty(converter, kAudioConverterEncodeBitRate, &propSize, &outputBitRate);
printf ("AAC Encode Bitrate: %u\n", (unsigned int)outputBitRate);
}
// can the Audio Converter resume conversion after an interruption?
// this property may be queried at any time after construction of the Audio Converter after setting its output format
// there's no clear reason to prefer construction time, interruption time, or potential resumption time but we prefer
// construction time since it means less code to execute during or after interruption time
UInt32 canResume = 0;
size = sizeof(canResume);
error = AudioConverterGetProperty(converter, kAudioConverterPropertyCanResumeFromInterruption, &size, &canResume);
if (noErr == error) {
// we recieved a valid return value from the GetProperty call
// if the property's value is 1, then the codec CAN resume work following an interruption
// if the property's value is 0, then interruptions destroy the codec's state and we're done
if (0 == canResume) canResumeFromInterruption = false;
printf("Audio Converter %s continue after interruption!\n", (canResumeFromInterruption == 0 ? "CANNOT" : "CAN"));
} else {
// if the property is unimplemented (kAudioConverterErr_PropertyNotSupported, or paramErr returned in the case of PCM),
// then the codec being used is not a hardware codec so we're not concerned about codec state
// we are always going to be able to resume conversion after an interruption
if (kAudioConverterErr_PropertyNotSupported == error) {
printf("kAudioConverterPropertyCanResumeFromInterruption property not supported - see comments in source for more info.\n");
} else {
printf("AudioConverterGetProperty kAudioConverterPropertyCanResumeFromInterruption result %ld, paramErr is OK if PCM\n", error);
}
error = noErr;
}
// create the destination file
XThrowIfError(AudioFileCreateWithURL(destinationURL, kAudioFileCAFType, &dstFormat, kAudioFileFlags_EraseFile, &destinationFileID), "AudioFileCreateWithURL failed!");
// set up source buffers and data proc info struct
afio.srcFileID = sourceFileID;
afio.srcBufferSize = 32768;
afio.srcBuffer = new char [afio.srcBufferSize];
afio.srcFilePos = 0;
afio.srcFormat = srcFormat;
if (srcFormat.mBytesPerPacket == 0) {
// if the source format is VBR, we need to get the maximum packet size
// use kAudioFilePropertyPacketSizeUpperBound which returns the theoretical maximum packet size
// in the file (without actually scanning the whole file to find the largest packet,
// as may happen with kAudioFilePropertyMaximumPacketSize)
size = sizeof(afio.srcSizePerPacket);
XThrowIfError(AudioFileGetProperty(sourceFileID, kAudioFilePropertyPacketSizeUpperBound, &size, &afio.srcSizePerPacket), "AudioFileGetProperty kAudioFilePropertyPacketSizeUpperBound failed!");
// how many packets can we read for our buffer size?
afio.numPacketsPerRead = afio.srcBufferSize / afio.srcSizePerPacket;
// allocate memory for the PacketDescription structures describing the layout of each packet
afio.packetDescriptions = new AudioStreamPacketDescription [afio.numPacketsPerRead];
} else {
// CBR source format
afio.srcSizePerPacket = srcFormat.mBytesPerPacket;
afio.numPacketsPerRead = afio.srcBufferSize / afio.srcSizePerPacket;
afio.packetDescriptions = NULL;
}
// set up output buffers
UInt32 outputSizePerPacket = dstFormat.mBytesPerPacket; // this will be non-zero if the format is CBR
UInt32 theOutputBufSize = 32768;
outputBuffer = new char[theOutputBufSize];
if (outputSizePerPacket == 0) {
// if the destination format is VBR, we need to get max size per packet from the converter
size = sizeof(outputSizePerPacket);
XThrowIfError(AudioConverterGetProperty(converter, kAudioConverterPropertyMaximumOutputPacketSize, &size, &outputSizePerPacket), "AudioConverterGetProperty kAudioConverterPropertyMaximumOutputPacketSize failed!");
// allocate memory for the PacketDescription structures describing the layout of each packet
outputPacketDescriptions = new AudioStreamPacketDescription [theOutputBufSize / outputSizePerPacket];
}
UInt32 numOutputPackets = theOutputBufSize / outputSizePerPacket;
// if the destination format has a cookie, get it and set it on the output file
WriteCookie(converter, destinationFileID);
// write destination channel layout
if (srcFormat.mChannelsPerFrame > 2) {
WriteDestinationChannelLayout(converter, sourceFileID, destinationFileID);
}
UInt64 totalOutputFrames = 0; // used for debgging printf
SInt64 outputFilePos = 0;
// loop to convert data
printf("Converting...\n");
while (1) {
// set up output buffer list
AudioBufferList fillBufList;
fillBufList.mNumberBuffers = 1;
fillBufList.mBuffers[0].mNumberChannels = dstFormat.mChannelsPerFrame;
fillBufList.mBuffers[0].mDataByteSize = theOutputBufSize;
fillBufList.mBuffers[0].mData = outputBuffer;
// this will block if we're interrupted
Boolean wasInterrupted = NO;
if ((error || wasInterrupted) && (false == canResumeFromInterruption)) {
// this is our interruption termination condition
// an interruption has occured but the Audio Converter cannot continue
error = kMyAudioConverterErr_CannotResumeFromInterruptionError;
break;
}
// convert data
UInt32 ioOutputDataPackets = numOutputPackets;
printf("AudioConverterFillComplexBuffer...\n");
error = AudioConverterFillComplexBuffer(converter, EncoderDataProc, &afio, &ioOutputDataPackets, &fillBufList, outputPacketDescriptions);
// if interrupted in the process of the conversion call, we must handle the error appropriately
if (error) {
if (kAudioConverterErr_HardwareInUse == error) {
printf("Audio Converter returned kAudioConverterErr_HardwareInUse!\n");
} else {
XThrowIfError(error, "AudioConverterFillComplexBuffer error!");
}
} else {
if (ioOutputDataPackets == 0) {
// this is the EOF conditon
error = noErr;
break;
}
}
if (noErr == error) {
// write to output file
UInt32 inNumBytes = fillBufList.mBuffers[0].mDataByteSize;
XThrowIfError(AudioFileWritePackets(destinationFileID, false, inNumBytes, outputPacketDescriptions, outputFilePos, &ioOutputDataPackets, outputBuffer), "AudioFileWritePackets failed!");
printf("Convert Output: Write %lu packets at position %lld, size: %ld\n", ioOutputDataPackets, outputFilePos, inNumBytes);
// advance output file packet position
outputFilePos += ioOutputDataPackets;
if (dstFormat.mFramesPerPacket) {
// the format has constant frames per packet
totalOutputFrames += (ioOutputDataPackets * dstFormat.mFramesPerPacket);
} else if (outputPacketDescriptions != NULL) {
// variable frames per packet require doing this for each packet (adding up the number of sample frames of data in each packet)
for (UInt32 i = 0; i < ioOutputDataPackets; ++i)
totalOutputFrames += outputPacketDescriptions[i].mVariableFramesInPacket;
}
}
} // while
if (noErr == error) {
// write out any of the leading and trailing frames for compressed formats only
if (dstFormat.mBitsPerChannel == 0) {
// our output frame count should jive with
printf("Total number of output frames counted: %lld\n", totalOutputFrames);
WritePacketTableInfo(converter, destinationFileID);
}
// write the cookie again - sometimes codecs will update cookies at the end of a conversion
WriteCookie(converter, destinationFileID);
}
}
catch (CAXException e) {
char buf[256];
fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
error = e.mError;
}
// cleanup
if (converter) AudioConverterDispose(converter);
if (destinationFileID) AudioFileClose(destinationFileID);
if (sourceFileID) AudioFileClose(sourceFileID);
if (afio.srcBuffer) delete [] afio.srcBuffer;
if (afio.packetDescriptions) delete [] afio.packetDescriptions;
if (outputBuffer) delete [] outputBuffer;
if (outputPacketDescriptions) delete [] outputPacketDescriptions;
return error;
}
void CAAudioFileConverter::GenerateOutputFileName(const char *inputFilePath,
const CAStreamBasicDescription &inputFormat,
const CAStreamBasicDescription &outputFormat, OSType outputFileType,
char *outName)
{
struct stat sb;
char inputDir[256];
char inputBasename[256];
strcpy(inputDir, dirname(inputFilePath));
const char *infname = basename(inputFilePath);
const char *inext = strrchr(infname, '.');
if (inext == NULL) strcpy(inputBasename, infname);
else {
int n;
memcpy(inputBasename, infname, n = inext - infname);
inputBasename[n] = '\0';
}
CFArrayRef exts;
UInt32 propSize = sizeof(exts);
XThrowIfError(AudioFileGetGlobalInfo(kAudioFileGlobalInfo_ExtensionsForType,
sizeof(OSType), &outputFileType, &propSize, &exts), "generate output file name");
char outputExt[32];
CFStringRef cfext = (CFStringRef)CFArrayGetValueAtIndex(exts, 0);
CFStringGetCString(cfext, outputExt, sizeof(outputExt), kCFStringEncodingUTF8);
CFRelease(exts);
// 1. olddir + oldname + newext
sprintf(outName, "%s/%s.%s", inputDir, inputBasename, outputExt);
#if TARGET_OS_MAC
if (lstat(outName, &sb)) return;
#else
if (stat(outName, &sb)) return;
#endif
if (outputFormat.IsPCM()) {
// If sample rate changed:
// 2. olddir + oldname + "-SR" + newext
if (inputFormat.mSampleRate != outputFormat.mSampleRate && outputFormat.mSampleRate != 0.) {
sprintf(outName, "%s/%s-%.0fk.%s", inputDir, inputBasename, outputFormat.mSampleRate/1000., outputExt);
#if TARGET_OS_MAC
if (lstat(outName, &sb)) return;
#else
if (stat(outName, &sb)) return;
#endif
}
// If bit depth changed:
// 3. olddir + oldname + "-bit" + newext
if (inputFormat.mBitsPerChannel != outputFormat.mBitsPerChannel) {
sprintf(outName, "%s/%s-%ldbit.%s", inputDir, inputBasename, outputFormat.mBitsPerChannel, outputExt);
#if TARGET_OS_MAC
if (lstat(outName, &sb)) return;
#else
if (stat(outName, &sb)) return;
#endif
}
}
// maybe more with channels/layouts? $$$
// now just append digits
for (int i = 1; ; ++i) {
sprintf(outName, "%s/%s-%d.%s", inputDir, inputBasename, i, outputExt);
#if TARGET_OS_MAC
if (lstat(outName, &sb)) return;
#else
if (stat(outName, &sb)) return;
#endif
}
}
void CAAudioFileConverter::ConvertFile(const ConversionParameters &_params)
{
FSRef destFSRef;
UInt32 propertySize;
CAStreamBasicDescription destFormat;
CAAudioChannelLayout origSrcFileLayout, srcFileLayout, destFileLayout;
bool openedSourceFile = false, createdOutputFile = false;
mParams = _params;
mReadBuffer = NULL;
mReadPtrs = NULL;
CABufferList *writeBuffer = NULL;
CABufferList *writePtrs = NULL;
PrepareConversion();
try {
if (TaggedDecodingFromCAF())
ReadCAFInfo();
OpenInputFile();
openedSourceFile = true;
// get input file's format
const CAStreamBasicDescription &srcFormat = mSrcFile.GetFileDataFormat();
if (mParams.flags & kOpt_Verbose) {
printf("Input file: %s, %qd frames\n", mParams.input.filePath ? basename(mParams.input.filePath) : "?",
mSrcFile.GetNumberFrames());
}
mSrcFormat = srcFormat;
bool encoding = !destFormat.IsPCM();
bool decoding = !srcFormat.IsPCM();
// prepare output file's format
destFormat = mParams.output.dataFormat;
if (!encoding && destFormat.mSampleRate == 0.)
// on encode, it's OK to have a 0 sample rate; ExtAudioFile will get the SR from the converter and set it on the file.
// on decode or PCM->PCM, a sample rate of 0 is interpreted as using the source sample rate
destFormat.mSampleRate = srcFormat.mSampleRate;
// source channel layout
srcFileLayout = mSrcFile.GetFileChannelLayout();
origSrcFileLayout = srcFileLayout;
if (mParams.input.channelLayoutTag != 0) {
XThrowIf(AudioChannelLayoutTag_GetNumberOfChannels(mParams.input.channelLayoutTag)
!= srcFormat.mChannelsPerFrame, -1, "input channel layout has wrong number of channels for file");
srcFileLayout = CAAudioChannelLayout(mParams.input.channelLayoutTag);
mSrcFile.SetFileChannelLayout(srcFileLayout);
}
// destination channel layout
int outChannels = mParams.output.channels;
if (mParams.output.channelLayoutTag != 0) {
// use the one specified by caller, if any
destFileLayout = CAAudioChannelLayout(mParams.output.channelLayoutTag);
} else if (srcFileLayout.IsValid()) {
// otherwise, assume the same as the source, if any
destFileLayout = srcFileLayout;
}
if (destFileLayout.IsValid()) {
// the output channel layout specifies the number of output channels
if (outChannels != -1)
XThrowIf((unsigned)outChannels != destFileLayout.NumberChannels(), -1,
"output channel layout has wrong number of channels");
else
outChannels = destFileLayout.NumberChannels();
}
if (!(mParams.flags & kOpt_NoSanitizeOutputFormat)) {
// adjust the output format's channels; output.channels overrides the channels
if (outChannels == -1)
outChannels = srcFormat.mChannelsPerFrame;
if (outChannels > 0) {
destFormat.mChannelsPerFrame = outChannels;
destFormat.mBytesPerPacket *= outChannels;
destFormat.mBytesPerFrame *= outChannels;
}
// use AudioFormat API to clean up the output format
propertySize = sizeof(AudioStreamBasicDescription);
XThrowIfError(AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &propertySize, &destFormat),
"get destination format info");
}
OpenOutputFile(srcFormat, destFormat, destFSRef, destFileLayout);
createdOutputFile = true;
mDestFormat = destFormat;
// set up client formats
CAStreamBasicDescription srcClientFormat, destClientFormat;
{
CAAudioChannelLayout srcClientLayout, destClientLayout;
if (encoding) {
if (decoding) {
// transcoding
// XThrowIf(encoding && decoding, -1, "transcoding not currently supported");
if (srcFormat.mChannelsPerFrame > 2 || destFormat.mChannelsPerFrame > 2)
CAXException::Warning("Transcoding multichannel audio may not handle channel layouts correctly", 0);
srcClientFormat.SetCanonical(std::min(srcFormat.mChannelsPerFrame, destFormat.mChannelsPerFrame), true);
srcClientFormat.mSampleRate = std::max(srcFormat.mSampleRate, destFormat.mSampleRate);
mSrcFile.SetClientFormat(srcClientFormat, NULL);
destClientFormat = srcClientFormat;
} else {
// encoding
srcClientFormat = srcFormat;
destClientFormat = srcFormat;
}
// by here, destClientFormat will have a valid sample rate
destClientLayout = srcFileLayout.IsValid() ? srcFileLayout : destFileLayout;
mDestFile.SetClientFormat(destClientFormat, &destClientLayout);
} else {
// decoding or PCM->PCM
if (destFormat.mSampleRate == 0.)
destFormat.mSampleRate = srcFormat.mSampleRate;
destClientFormat = destFormat;
srcClientFormat = destFormat;
srcClientLayout = destFileLayout;
mSrcFile.SetClientFormat(srcClientFormat, &srcClientLayout);
}
}
XThrowIf(srcClientFormat.mBytesPerPacket == 0, -1, "source client format not PCM");
XThrowIf(destClientFormat.mBytesPerPacket == 0, -1, "dest client format not PCM");
if (encoding) {
// set the bitrate
if (mParams.output.bitRate != -1) {
if (mParams.flags & kOpt_Verbose)
printf("bitrate = %ld\n", mParams.output.bitRate);
mDestFile.SetConverterProperty(kAudioConverterEncodeBitRate, sizeof(UInt32), &mParams.output.bitRate);
}
// set the codec quality
if (mParams.output.codecQuality != -1) {
if (mParams.flags & kOpt_Verbose)
printf("codec quality = %ld\n", mParams.output.codecQuality);
mDestFile.SetConverterProperty(kAudioConverterCodecQuality, sizeof(UInt32), &mParams.output.codecQuality);
}
// set the bitrate strategy -- called bitrate format in the codecs since it had already shipped
if (mParams.output.strategy != -1) {
if (mParams.flags & kOpt_Verbose)
printf("strategy = %ld\n", mParams.output.strategy);
mDestFile.SetConverterProperty(kAudioCodecBitRateFormat, sizeof(UInt32), &mParams.output.strategy);
}
}
// set the SRC quality
if (mParams.output.srcQuality != -1) {
if (srcFormat.mSampleRate != 0. && destFormat.mSampleRate != 0. && srcFormat.mSampleRate != destFormat.mSampleRate) {
if (mParams.flags & kOpt_Verbose)
printf("SRC quality = %ld\n", mParams.output.srcQuality);
if (encoding)
mDestFile.SetConverterProperty(kAudioConverterSampleRateConverterQuality, sizeof(UInt32), &mParams.output.srcQuality);
else
mSrcFile.SetConverterProperty(kAudioConverterSampleRateConverterQuality, sizeof(UInt32), &mParams.output.srcQuality);
}
}
if (decoding) {
if (mParams.output.primeMethod != -1)
mSrcFile.SetConverterProperty(kAudioConverterPrimeMethod, sizeof(UInt32), &mParams.output.primeMethod);
}
PrintFormats(&origSrcFileLayout);
// prepare I/O buffers
UInt32 bytesToRead = 0x10000;
UInt32 framesToRead = bytesToRead; // OK, ReadPackets will limit as appropriate
ComputeReadSize(srcFormat, destFormat, bytesToRead, framesToRead);
// const SInt64 totalFrames = mSrcFile.GetNumberFrames();
//#warning "GetNumberFrames() can be prohibitively slow for some formats"
mReadBuffer = CABufferList::New("readbuf", srcClientFormat);
mReadBuffer->AllocateBuffers(bytesToRead);
mReadPtrs = CABufferList::New("readptrs", srcClientFormat);
BeginConversion();
while (true) {
//XThrowIf(Progress(mSrcFile.Tell(), totalFrames), userCanceledErr, "user stopped");
// this was commented out for awhile -- performance? make it optional?
UInt32 nFrames = framesToRead;
mReadPtrs->SetFrom(mReadBuffer);
AudioBufferList *readbuf = &mReadPtrs->GetModifiableBufferList();
mSrcFile.Read(nFrames, readbuf);
//printf("read %ld of %ld frames\n", nFrames, framesToRead);
if (nFrames == 0)
break;
mDestFile.Write(nFrames, readbuf);
if (ShouldTerminateConversion())
break;
}
if (decoding) {
// fix up the destination file's length if necessary and possible
SInt64 nframes = mSrcFile.GetNumberFrames();
if (nframes != 0) {
// only shorten, don't try to lengthen
nframes = SInt64(ceil(nframes * destFormat.mSampleRate / srcFormat.mSampleRate));
if (nframes < mDestFile.GetNumberFrames()) {
mDestFile.SetNumberFrames(nframes);
}
}
}
EndConversion();
}
catch (...) {
delete mReadBuffer;
delete mReadPtrs;
delete writeBuffer;
delete writePtrs;
if (!createdOutputFile)
PrintFormats(&origSrcFileLayout);
try { mSrcFile.Close(); } catch (...) { }
try { mDestFile.Close(); } catch (...) { }
if (createdOutputFile)
unlink(mOutName);
throw;
}
delete mReadBuffer;
delete mReadPtrs;
delete writeBuffer;
delete writePtrs;
mSrcFile.Close();
mDestFile.Close();
if (TaggedEncodingToCAF())
WriteCAFInfo();
if (mParams.flags & kOpt_Verbose) {
// must close to flush encoder; GetNumberFrames() not necessarily valid until afterwards but then
// the file is closed
CAAudioFile temp;
FSRef destFSRef;
if (FSPathMakeRef((UInt8 *)mOutName, &destFSRef, NULL) == noErr) {
temp.Open(destFSRef);
printf("Output file: %s, %qd frames\n", basename(mOutName), temp.GetNumberFrames());
}
}
}
OSStatus DoConvertFile(CFURLRef sourceURL, CFURLRef destinationURL, OSType outputFormat, Float64 outputSampleRate)
{
ExtAudioFileRef sourceFile = 0;
ExtAudioFileRef destinationFile = 0;
Boolean canResumeFromInterruption = true; // we can continue unless told otherwise
OSStatus error = noErr;
// in this sample we should never be on the main thread here
assert(![NSThread isMainThread]);
// transition thread state to kStateRunning before continuing
ThreadStateSetRunning();
printf("DoConvertFile\n");
try {
CAStreamBasicDescription srcFormat, dstFormat;
// open the source file
XThrowIfError(ExtAudioFileOpenURL(sourceURL, &sourceFile), "ExtAudioFileOpenURL failed");
// get the source data format
UInt32 size = sizeof(srcFormat);
XThrowIfError(ExtAudioFileGetProperty(sourceFile, kExtAudioFileProperty_FileDataFormat, &size, &srcFormat), "couldn't get source data format");
printf("\nSource file format: "); srcFormat.Print();
// setup the output file format
dstFormat.mSampleRate = (outputSampleRate == 0 ? srcFormat.mSampleRate : outputSampleRate); // set sample rate
if (outputFormat == kAudioFormatLinearPCM) {
// if PCM was selected as the destination format, create a 16-bit int PCM file format description
dstFormat.mFormatID = outputFormat;
dstFormat.mChannelsPerFrame = srcFormat.NumberChannels();
dstFormat.mBitsPerChannel = 16;
dstFormat.mBytesPerPacket = dstFormat.mBytesPerFrame = 2 * dstFormat.mChannelsPerFrame;
dstFormat.mFramesPerPacket = 1;
dstFormat.mFormatFlags = kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger; // little-endian
} else {
// compressed format - need to set at least format, sample rate and channel fields for kAudioFormatProperty_FormatInfo
dstFormat.mFormatID = outputFormat;
dstFormat.mChannelsPerFrame = (outputFormat == kAudioFormatiLBC ? 1 : srcFormat.NumberChannels()); // for iLBC num channels must be 1
// use AudioFormat API to fill out the rest of the description
size = sizeof(dstFormat);
XThrowIfError(AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &dstFormat), "couldn't create destination data format");
}
printf("\nDestination file format: "); dstFormat.Print();
// create the destination file
XThrowIfError(ExtAudioFileCreateWithURL(destinationURL, kAudioFileCAFType, &dstFormat, NULL, kAudioFileFlags_EraseFile, &destinationFile), "ExtAudioFileCreateWithURL failed!");
// set the client format - The format must be linear PCM (kAudioFormatLinearPCM)
// You must set this in order to encode or decode a non-PCM file data format
// You may set this on PCM files to specify the data format used in your calls to read/write
CAStreamBasicDescription clientFormat;
if (outputFormat == kAudioFormatLinearPCM) {
clientFormat = dstFormat;
} else {
clientFormat.SetCanonical(srcFormat.NumberChannels(), true);
clientFormat.mSampleRate = srcFormat.mSampleRate;
}
printf("\nClient data format: "); clientFormat.Print();
printf("\n");
size = sizeof(clientFormat);
XThrowIfError(ExtAudioFileSetProperty(sourceFile, kExtAudioFileProperty_ClientDataFormat, size, &clientFormat), "couldn't set source client format");
size = sizeof(clientFormat);
XThrowIfError(ExtAudioFileSetProperty(destinationFile, kExtAudioFileProperty_ClientDataFormat, size, &clientFormat), "couldn't set destination client format");
// can the audio converter (which in this case is owned by an ExtAudioFile object) resume conversion after an interruption?
AudioConverterRef audioConverter;
size = sizeof(audioConverter);
XThrowIfError(ExtAudioFileGetProperty(destinationFile, kExtAudioFileProperty_AudioConverter, &size, &audioConverter), "Couldn't get Audio Converter!");
// this property may be queried at any time after construction of the audio converter (which in this case is owned by an ExtAudioFile object)
// after setting the output format -- there's no clear reason to prefer construction time, interruption time, or potential resumption time but we prefer
// construction time since it means less code to execute during or after interruption time
UInt32 canResume = 0;
size = sizeof(canResume);
error = AudioConverterGetProperty(audioConverter, kAudioConverterPropertyCanResumeFromInterruption, &size, &canResume);
if (noErr == error) {
// we recieved a valid return value from the GetProperty call
// if the property's value is 1, then the codec CAN resume work following an interruption
// if the property's value is 0, then interruptions destroy the codec's state and we're done
if (0 == canResume) canResumeFromInterruption = false;
printf("Audio Converter %s continue after interruption!\n", (canResumeFromInterruption == 0 ? "CANNOT" : "CAN"));
} else {
// if the property is unimplemented (kAudioConverterErr_PropertyNotSupported, or paramErr returned in the case of PCM),
// then the codec being used is not a hardware codec so we're not concerned about codec state
// we are always going to be able to resume conversion after an interruption
if (kAudioConverterErr_PropertyNotSupported == error) {
printf("kAudioConverterPropertyCanResumeFromInterruption property not supported!\n");
} else {
printf("AudioConverterGetProperty kAudioConverterPropertyCanResumeFromInterruption result %ld\n", error);
}
error = noErr;
}
// set up buffers
UInt32 bufferByteSize = 32768;
char srcBuffer[bufferByteSize];
// keep track of the source file offset so we know where to reset the source for
// reading if interrupted and input was not consumed by the audio converter
SInt64 sourceFrameOffset = 0;
//***** do the read and write - the conversion is done on and by the write call *****//
printf("Converting...\n");
while (1) {
AudioBufferList fillBufList;
fillBufList.mNumberBuffers = 1;
fillBufList.mBuffers[0].mNumberChannels = clientFormat.NumberChannels();
fillBufList.mBuffers[0].mDataByteSize = bufferByteSize;
fillBufList.mBuffers[0].mData = srcBuffer;
// client format is always linear PCM - so here we determine how many frames of lpcm
// we can read/write given our buffer size
UInt32 numFrames;
if (clientFormat.mBytesPerFrame > 0) // rids bogus analyzer div by zero warning mBytesPerFrame can't be 0 and is protected by an Assert
numFrames = clientFormat.BytesToFrames(bufferByteSize); // (bufferByteSize / clientFormat.mBytesPerFrame);
XThrowIfError(ExtAudioFileRead(sourceFile, &numFrames, &fillBufList), "ExtAudioFileRead failed!");
if (!numFrames) {
// this is our termination condition
error = noErr;
break;
}
sourceFrameOffset += numFrames;
// this will block if we're interrupted
Boolean wasInterrupted = ThreadStatePausedCheck();
if ((error || wasInterrupted) && (false == canResumeFromInterruption)) {
// this is our interruption termination condition
// an interruption has occured but the audio converter cannot continue
error = kMyAudioConverterErr_CannotResumeFromInterruptionError;
break;
}
error = ExtAudioFileWrite(destinationFile, numFrames, &fillBufList);
// if interrupted in the process of the write call, we must handle the errors appropriately
if (error) {
if (kExtAudioFileError_CodecUnavailableInputConsumed == error) {
printf("ExtAudioFileWrite kExtAudioFileError_CodecUnavailableInputConsumed error %ld\n", error);
/*
Returned when ExtAudioFileWrite was interrupted. You must stop calling
ExtAudioFileWrite. If the underlying audio converter can resume after an
interruption (see kAudioConverterPropertyCanResumeFromInterruption), you must
wait for an EndInterruption notification from AudioSession, then activate the session
before resuming. In this situation, the buffer you provided to ExtAudioFileWrite was successfully
consumed and you may proceed to the next buffer
*/
} else if (kExtAudioFileError_CodecUnavailableInputNotConsumed == error) {
printf("ExtAudioFileWrite kExtAudioFileError_CodecUnavailableInputNotConsumed error %ld\n", error);
/*
Returned when ExtAudioFileWrite was interrupted. You must stop calling
ExtAudioFileWrite. If the underlying audio converter can resume after an
interruption (see kAudioConverterPropertyCanResumeFromInterruption), you must
wait for an EndInterruption notification from AudioSession, then activate the session
before resuming. In this situation, the buffer you provided to ExtAudioFileWrite was not
successfully consumed and you must try to write it again
*/
// seek back to last offset before last read so we can try again after the interruption
sourceFrameOffset -= numFrames;
XThrowIfError(ExtAudioFileSeek(sourceFile, sourceFrameOffset), "ExtAudioFileSeek failed!");
} else {
XThrowIfError(error, "ExtAudioFileWrite error!");
}
} // if
} // while
}
catch (CAXException e) {
char buf[256];
fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
error = e.mError;
}
// close
if (destinationFile) ExtAudioFileDispose(destinationFile);
if (sourceFile) ExtAudioFileDispose(sourceFile);
// transition thread state to kStateDone before continuing
ThreadStateSetDone();
return error;
}
int main (int argc, const char * argv[])
{
#if TARGET_OS_WIN32
InitializeQTML(0L);
#endif
const char *fpath = NULL;
Float32 volume = 1;
Float32 duration = -1;
Float32 currentTime = 0.0;
Float32 rate = 0;
int rQuality = 0;
bool doPrint = false;
for (int i = 1; i < argc; ++i) {
const char *arg = argv[i];
if (arg[0] != '-') {
if (fpath != NULL) {
fprintf(stderr, "may only specify one file to play\n");
usage();
}
fpath = arg;
} else {
arg += 1;
if (arg[0] == 'v' || !strcmp(arg, "-volume")) {
if (++i == argc)
MissingArgument();
arg = argv[i];
sscanf(arg, "%f", &volume);
} else if (arg[0] == 't' || !strcmp(arg, "-time")) {
if (++i == argc)
MissingArgument();
arg = argv[i];
sscanf(arg, "%f", &duration);
} else if (arg[0] == 'r' || !strcmp(arg, "-rate")) {
if (++i == argc)
MissingArgument();
arg = argv[i];
sscanf(arg, "%f", &rate);
} else if (arg[0] == 'q' || !strcmp(arg, "-rQuality")) {
if (++i == argc)
MissingArgument();
arg = argv[i];
sscanf(arg, "%d", &rQuality);
} else if (arg[0] == 'h' || !strcmp(arg, "-help")) {
usage();
} else if (arg[0] == 'd' || !strcmp(arg, "-debug")) {
doPrint = true;
} else {
fprintf(stderr, "unknown argument: %s\n\n", arg - 1);
usage();
}
}
}
if (fpath == NULL)
usage();
if (doPrint)
printf ("Playing file: %s\n", fpath);
try {
AQTestInfo myInfo;
CFURLRef sndFile = CFURLCreateFromFileSystemRepresentation (NULL, (const UInt8 *)fpath, strlen(fpath), false);
if (!sndFile) XThrowIfError (!sndFile, "can't parse file path");
OSStatus result = AudioFileOpenURL (sndFile, 0x1/*fsRdPerm*/, 0/*inFileTypeHint*/, &myInfo.mAudioFile);
CFRelease (sndFile);
XThrowIfError(result, "AudioFileOpen failed");
UInt32 size;
XThrowIfError(AudioFileGetPropertyInfo(myInfo.mAudioFile,
kAudioFilePropertyFormatList, &size, NULL), "couldn't get file's format list info");
UInt32 numFormats = size / sizeof(AudioFormatListItem);
AudioFormatListItem *formatList = new AudioFormatListItem [ numFormats ];
XThrowIfError(AudioFileGetProperty(myInfo.mAudioFile,
kAudioFilePropertyFormatList, &size, formatList), "couldn't get file's data format");
numFormats = size / sizeof(AudioFormatListItem); // we need to reassess the actual number of formats when we get it
if (numFormats == 1) {
// this is the common case
myInfo.mDataFormat = formatList[0].mASBD;
// see if there is a channel layout (multichannel file)
result = AudioFileGetPropertyInfo(myInfo.mAudioFile, kAudioFilePropertyChannelLayout, &myInfo.mChannelLayoutSize, NULL);
if (result == noErr && myInfo.mChannelLayoutSize > 0) {
myInfo.mChannelLayout = (AudioChannelLayout *)new char [myInfo.mChannelLayoutSize];
XThrowIfError(AudioFileGetProperty(myInfo.mAudioFile, kAudioFilePropertyChannelLayout, &myInfo.mChannelLayoutSize, myInfo.mChannelLayout), "get audio file's channel layout");
}
} else {
if (doPrint) {
printf ("File has a %d layered data format:\n", (int)numFormats);
for (unsigned int i = 0; i < numFormats; ++i)
CAStreamBasicDescription(formatList[i].mASBD).Print();
}
// now we should look to see which decoders we have on the system
XThrowIfError(AudioFormatGetPropertyInfo(kAudioFormatProperty_DecodeFormatIDs, 0, NULL, &size), "couldn't get decoder id's");
UInt32 numDecoders = size / sizeof(OSType);
OSType *decoderIDs = new OSType [ numDecoders ];
XThrowIfError(AudioFormatGetProperty(kAudioFormatProperty_DecodeFormatIDs, 0, NULL, &size, decoderIDs), "couldn't get decoder id's");
unsigned int i = 0;
for (; i < numFormats; ++i) {
OSType decoderID = formatList[i].mASBD.mFormatID;
bool found = false;
for (unsigned int j = 0; j < numDecoders; ++j) {
if (decoderID == decoderIDs[j]) {
found = true;
break;
}
}
if (found) break;
}
delete [] decoderIDs;
if (i >= numFormats) {
fprintf (stderr, "Cannot play any of the formats in this file\n");
throw kAudioFileUnsupportedDataFormatError;
}
myInfo.mDataFormat = formatList[i].mASBD;
myInfo.mChannelLayoutSize = sizeof(AudioChannelLayout);
myInfo.mChannelLayout = (AudioChannelLayout*)new char [myInfo.mChannelLayoutSize];
myInfo.mChannelLayout->mChannelLayoutTag = formatList[i].mChannelLayoutTag;
myInfo.mChannelLayout->mChannelBitmap = 0;
myInfo.mChannelLayout->mNumberChannelDescriptions = 0;
}
delete [] formatList;
if (doPrint) {
printf ("Playing format: ");
myInfo.mDataFormat.Print();
}
XThrowIfError(AudioQueueNewOutput(&myInfo.mDataFormat, AQTestBufferCallback, &myInfo,
CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &myInfo.mQueue), "AudioQueueNew failed");
UInt32 bufferByteSize;
// we need to calculate how many packets we read at a time, and how big a buffer we need
// we base this on the size of the packets in the file and an approximate duration for each buffer
{
bool isFormatVBR = (myInfo.mDataFormat.mBytesPerPacket == 0 || myInfo.mDataFormat.mFramesPerPacket == 0);
// first check to see what the max size of a packet is - if it is bigger
// than our allocation default size, that needs to become larger
UInt32 maxPacketSize;
size = sizeof(maxPacketSize);
XThrowIfError(AudioFileGetProperty(myInfo.mAudioFile,
kAudioFilePropertyPacketSizeUpperBound, &size, &maxPacketSize), "couldn't get file's max packet size");
// adjust buffer size to represent about a half second of audio based on this format
CalculateBytesForTime (myInfo.mDataFormat, maxPacketSize, 0.5/*seconds*/, &bufferByteSize, &myInfo.mNumPacketsToRead);
if (isFormatVBR)
myInfo.mPacketDescs = new AudioStreamPacketDescription [myInfo.mNumPacketsToRead];
else
myInfo.mPacketDescs = NULL; // we don't provide packet descriptions for constant bit rate formats (like linear PCM)
if (doPrint)
printf ("Buffer Byte Size: %d, Num Packets to Read: %d\n", (int)bufferByteSize, (int)myInfo.mNumPacketsToRead);
}
// (2) If the file has a cookie, we should get it and set it on the AQ
size = sizeof(UInt32);
result = AudioFileGetPropertyInfo (myInfo.mAudioFile, kAudioFilePropertyMagicCookieData, &size, NULL);
if (!result && size) {
char* cookie = new char [size];
XThrowIfError (AudioFileGetProperty (myInfo.mAudioFile, kAudioFilePropertyMagicCookieData, &size, cookie), "get cookie from file");
XThrowIfError (AudioQueueSetProperty(myInfo.mQueue, kAudioQueueProperty_MagicCookie, cookie, size), "set cookie on queue");
delete [] cookie;
}
// set ACL if there is one
if (myInfo.mChannelLayout)
XThrowIfError(AudioQueueSetProperty(myInfo.mQueue, kAudioQueueProperty_ChannelLayout, myInfo.mChannelLayout, myInfo.mChannelLayoutSize), "set channel layout on queue");
// prime the queue with some data before starting
myInfo.mDone = false;
myInfo.mCurrentPacket = 0;
for (int i = 0; i < kNumberBuffers; ++i) {
XThrowIfError(AudioQueueAllocateBuffer(myInfo.mQueue, bufferByteSize, &myInfo.mBuffers[i]), "AudioQueueAllocateBuffer failed");
AQTestBufferCallback (&myInfo, myInfo.mQueue, myInfo.mBuffers[i]);
if (myInfo.mDone) break;
}
// set the volume of the queue
XThrowIfError (AudioQueueSetParameter(myInfo.mQueue, kAudioQueueParam_Volume, volume), "set queue volume");
XThrowIfError (AudioQueueAddPropertyListener (myInfo.mQueue, kAudioQueueProperty_IsRunning, MyAudioQueuePropertyListenerProc, NULL), "add listener");
#if !TARGET_OS_IPHONE
if (rate > 0) {
UInt32 propValue = 1;
XThrowIfError (AudioQueueSetProperty (myInfo.mQueue, kAudioQueueProperty_EnableTimePitch, &propValue, sizeof(propValue)), "enable time pitch");
propValue = rQuality ? kAudioQueueTimePitchAlgorithm_Spectral : kAudioQueueTimePitchAlgorithm_TimeDomain;
XThrowIfError (AudioQueueSetProperty (myInfo.mQueue, kAudioQueueProperty_TimePitchAlgorithm, &propValue, sizeof(propValue)), "time pitch algorithm");
propValue = (rate == 1.0f ? 1 : 0); // bypass rate if 1.0
XThrowIfError (AudioQueueSetProperty (myInfo.mQueue, kAudioQueueProperty_TimePitchBypass, &propValue, sizeof(propValue)), "bypass time pitch");
if (rate != 1) {
XThrowIfError (AudioQueueSetParameter (myInfo.mQueue, kAudioQueueParam_PlayRate, rate), "set playback rate");
}
if (doPrint) {
printf ("Enable rate-scaled playback (rate = %.2f) using %s algorithm\n", rate, (rQuality ? "Spectral": "Time Domain"));
}
}
#endif
// lets start playing now - stop is called in the AQTestBufferCallback when there's
// no more to read from the file
XThrowIfError(AudioQueueStart(myInfo.mQueue, NULL), "AudioQueueStart failed");
do {
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.25, false);
currentTime += .25;
if (duration > 0 && currentTime >= duration)
break;
} while (gIsRunning);
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 1, false);
XThrowIfError(AudioQueueDispose(myInfo.mQueue, true), "AudioQueueDispose(true) failed");
XThrowIfError(AudioFileClose(myInfo.mAudioFile), "AudioQueueDispose(false) failed");
}
catch (CAXException e) {
char buf[256];
fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
}
catch (...) {
fprintf(stderr, "Unspecified exception\n");
}
return 0;
}
void CAChannelMappingPlayer::SetupChannelMapping()
{
delete mMapper;
mMapper = NULL;
const CAStreamBasicDescription &fileFormat = GetFile().GetClientDataFormat();
CAStreamBasicDescription deviceFormat;
UInt32 propertySize = sizeof(AudioStreamBasicDescription);
XThrowIfError(AudioUnitGetProperty(
GetOutputUnit(),
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
0,
(void *)&deviceFormat,
&propertySize), "get output device's format");
#if VERBOSE
printf("CAChannelMappingPlayer::SetupChannelMapping: %ld-ch file, %ld-ch device\n",
fileFormat.mChannelsPerFrame, deviceFormat.mChannelsPerFrame);
#endif
if (fileFormat.mChannelsPerFrame <= deviceFormat.mChannelsPerFrame) {
// no mapping needed, use output unit's default behavior
// (default stereo pair and speaker config from AMS)
#if VERBOSE
printf(" using output unit's channel mapping\n");
#endif
CAAudioFilePlayer::SetupChannelMapping();
} else {
// fewer device than file channels, mapping needed
CAAudioChannelLayout fileLayout, deviceLayout;
#if VERBOSE
printf(" using our own channel mapping\n");
#endif
deviceFormat.mSampleRate = fileFormat.mSampleRate;
deviceFormat.SetCanonical(deviceFormat.mChannelsPerFrame, false); // force deinterleaved
fileLayout = GetFile().GetFileChannelLayout();
UInt32 layoutSize;
Boolean writable;
OSStatus err = AudioUnitGetPropertyInfo(
GetOutputUnit(),
kAudioUnitProperty_AudioChannelLayout,
kAudioUnitScope_Input,
0,
&layoutSize,
&writable);
if (!err) {
char *buf = (char *)malloc(layoutSize);
err = AudioUnitGetProperty(
GetOutputUnit(),
kAudioUnitProperty_AudioChannelLayout,
kAudioUnitScope_Input,
0,
buf,
&layoutSize);
deviceLayout = CAAudioChannelLayout(reinterpret_cast<AudioChannelLayout *>(buf));
free(buf);
}
mMapper = new CAChannelMapper(fileFormat, deviceFormat, &fileLayout, &deviceLayout);
// give the output unit the same number of channels as in the device,
// since we'll be doing the mapping ourselves
XThrowIfError(AudioUnitSetProperty(
GetOutputUnit(),
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
0,
(void *)&deviceFormat,
sizeof(AudioStreamBasicDescription)), "set audio output format");
XThrowIfError(mMapper->OpenMixer(fileFormat.mSampleRate), "open mixer");
XThrowIfError(mMapper->ConfigureDownmix(), "configure downmix");
AudioUnitConnection conn;
conn.sourceAudioUnit = mMapper->GetMixer();
conn.sourceOutputNumber = 0;
conn.destInputNumber = 0;
XThrowIfError(AudioUnitSetProperty(
GetOutputUnit(),
kAudioUnitProperty_MakeConnection,
kAudioUnitScope_Global,
0,
(void *)&conn,
sizeof(AudioUnitConnection)), "connect mixer to output unit");
AURenderCallbackStruct input;
input.inputProc = InputProc;
input.inputProcRefCon = this;
XThrowIfError(AudioUnitSetProperty(
conn.sourceAudioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
0,
&input,
sizeof(input)), "connect input proc to mixer");
// provide NO channel layout
// mReadBuf = CABufferList::New("", fileFormat);
// mReadBuf->AllocateBuffers(
}
}
