// _______________________________________________________________________________________
//
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
}
}
// ____________________________________________________________________________________
// 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;
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
CheckError(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)
CheckError(AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL), "AudioQueueEnqueueBuffer failed");
}
// ____________________________________________________________________________________
// 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));
}
}
// Audio Queue callback function, called when an input buffer has been filled.
static void MyAQInputCallback(void *inUserData, AudioQueueRef inQueue,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp *inStartTime,
UInt32 inNumPackets,
const AudioStreamPacketDescription *inPacketDesc)
{
MyRecorder *recorder = (MyRecorder *)inUserData;
// if inNumPackets is greater then zero, our buffer contains audio data
// in the format we specified (AAC)
if (inNumPackets > 0)
{
// write packets to file
CheckError(AudioFileWritePackets(recorder->recordFile, FALSE, inBuffer->mAudioDataByteSize,
inPacketDesc, recorder->recordPacket, &inNumPackets,
inBuffer->mAudioData), "AudioFileWritePackets failed");
// increment packet index
recorder->recordPacket += inNumPackets;
}
// if we're not stopping, re-enqueue the buffer so that it gets filled again
if (recorder->running)
CheckError(AudioQueueEnqueueBuffer(inQueue, inBuffer,
0, NULL), "AudioQueueEnqueueBuffer failed");
}
void HLAudioFile::WriteAudioFrames(SInt64 inOffset, UInt32& ioNumberFrames, void* inData, bool inCache)
{
ThrowIf(mAudioFileID == 0, CAException(fnOpnErr), "HLAudioFile::WriteAudioFrames: file isn't prepared");
UInt32 theNumberBytesToWrite = ioNumberFrames * mFormat.mBytesPerFrame;
OSStatus theError = AudioFileWritePackets(mAudioFileID, inCache, theNumberBytesToWrite, NULL, inOffset, &ioNumberFrames, inData);
ThrowIfError(theError, CAException(theError), "HLAudioFile::WriteAudioFrames: couldn't write the data");
}
void Convert(MyAudioConverterSettings *mySettings)
{
UInt32 outputBufferSize = 32 * 1024; // 32 KB is a good starting point
UInt32 sizePerPacket = mySettings->outputFormat.mBytesPerPacket;
UInt32 packetsPerBuffer = outputBufferSize / sizePerPacket;
// allocate destination buffer
UInt8 *outputBuffer = (UInt8 *)malloc(sizeof(UInt8) * outputBufferSize);
UInt32 outputFilePacketPosition = 0; //in bytes
while(1)
{
// wrap the destination buffer in an AudioBufferList
AudioBufferList convertedData;
convertedData.mNumberBuffers = 1;
convertedData.mBuffers[0].mNumberChannels = mySettings->outputFormat.mChannelsPerFrame;
convertedData.mBuffers[0].mDataByteSize = outputBufferSize;
convertedData.mBuffers[0].mData = outputBuffer;
UInt32 frameCount = packetsPerBuffer;
// read from the extaudiofile
CheckResult(ExtAudioFileRead(mySettings->inputFile,
&frameCount,
&convertedData),
"Couldn't read from input file");
if (frameCount == 0) {
printf ("done reading from file");
return;
}
// write the converted data to the output file
CheckResult (AudioFileWritePackets(mySettings->outputFile,
FALSE,
frameCount,
NULL,
outputFilePacketPosition / mySettings->outputFormat.mBytesPerPacket,
&frameCount,
convertedData.mBuffers[0].mData),
"Couldn't write packets to file");
// advance the output file write location
outputFilePacketPosition += (frameCount * mySettings->outputFormat.mBytesPerPacket);
}
// AudioConverterDispose(audioConverter);
}
// _______________________________________________________________________________________
//
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;
}
}
void SoundRecorder::handleInputBuffer(AudioQueueRef inAQ,AudioQueueBufferRef inBuffer,const AudioTimeStamp* inStartTime,UInt32 inNumPackets,const AudioStreamPacketDescription* inPacketDesc)
{
/* Calculate the number of packets in the buffer if not given: */
if(inNumPackets==0&&format.mBytesPerPacket!=0)
inNumPackets=inBuffer->mAudioDataByteSize/format.mBytesPerPacket;
/* Write the just-filled buffer to the audio file: */
if(AudioFileWritePackets(audioFile,false,inBuffer->mAudioDataByteSize,inPacketDesc,numRecordedPackets,&inNumPackets,inBuffer->mAudioData)==noErr)
{
/* Update the number of written packets: */
numRecordedPackets+=inNumPackets;
}
/* Put the just-filled buffer back into the audio queue if we're still recording: */
if(active)
AudioQueueEnqueueBuffer(queue,inBuffer,0,0);
}
void DiskOutUGenInternal::writeBuffer(const int bufferSize) throw()
{
UInt32 ioNumPackets = bufferSize;
OSStatus status = AudioFileWritePackets(audioFile_, false,
allocatedBlockSizeInBytes, NULL,
currentPacket,&ioNumPackets,
audioData);
if((status != noErr) || (ioNumPackets == 0))
{
printf("DiskOut: error: writing file\n");
AudioFileClose(audioFile_);
audioFile_ = 0;
}
else
{
currentPacket += ioNumPackets;
}
}
void HandleInputBuffer (
void *aqData,
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp *inStartTime,
UInt32 inNumPackets,
const AudioStreamPacketDescription *inPacketDesc
) {
AQRecorderState *pAqData = (AQRecorderState *) aqData; // 1
if (inNumPackets == 0 && // 2
pAqData->mDataFormat.mBytesPerPacket != 0)
inNumPackets =
inBuffer->mAudioDataByteSize / pAqData->mDataFormat.mBytesPerPacket;
if (AudioFileWritePackets ( // 3
pAqData->mAudioFile,
false,
inBuffer->mAudioDataByteSize,
inPacketDesc,
pAqData->mCurrentPacket,
&inNumPackets,
inBuffer->mAudioData
) == noErr) {
pAqData->mCurrentPacket += inNumPackets;} // 4
if (pAqData->mIsRunning == 0) // 5
return;
AudioQueueEnqueueBuffer ( // 6
pAqData->mQueue,
inBuffer,
0,
NULL
);
}
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 Convert(MyAudioConverterSettings *mySettings)
{
// create audioConverter object
AudioConverterRef audioConverter;
CheckResult (AudioConverterNew(&mySettings->inputFormat, &mySettings->outputFormat, &audioConverter),
"AudioConveterNew failed");
// allocate packet descriptions if the input file is VBR
UInt32 packetsPerBuffer = 0;
UInt32 outputBufferSize = 32 * 1024; // 32 KB is a good starting point
UInt32 sizePerPacket = mySettings->inputFormat.mBytesPerPacket;
if (sizePerPacket == 0)
{
UInt32 size = sizeof(sizePerPacket);
CheckResult(AudioConverterGetProperty(audioConverter, kAudioConverterPropertyMaximumOutputPacketSize, &size, &sizePerPacket),
"Couldn't get kAudioConverterPropertyMaximumOutputPacketSize");
// make sure the buffer is large enough to hold at least one packet
if (sizePerPacket > outputBufferSize)
outputBufferSize = sizePerPacket;
packetsPerBuffer = outputBufferSize / sizePerPacket;
mySettings->inputFilePacketDescriptions = (AudioStreamPacketDescription*)malloc(sizeof(AudioStreamPacketDescription) * packetsPerBuffer);
}
else
{
packetsPerBuffer = outputBufferSize / sizePerPacket;
}
// allocate destination buffer
UInt8 *outputBuffer = (UInt8 *)malloc(sizeof(UInt8) * outputBufferSize); // CHRIS: not sizeof(UInt8*). check book text!
UInt32 outputFilePacketPosition = 0; //in bytes
while(1)
{
// wrap the destination buffer in an AudioBufferList
AudioBufferList convertedData;
convertedData.mNumberBuffers = 1;
convertedData.mBuffers[0].mNumberChannels = mySettings->inputFormat.mChannelsPerFrame;
convertedData.mBuffers[0].mDataByteSize = outputBufferSize;
convertedData.mBuffers[0].mData = outputBuffer;
// now call the audioConverter to transcode the data. This function will call
// the callback function as many times as required to fulfill the request.
UInt32 ioOutputDataPackets = packetsPerBuffer;
OSStatus error = AudioConverterFillComplexBuffer(audioConverter,
MyAudioConverterCallback,
mySettings,
&ioOutputDataPackets,
&convertedData,
(mySettings->inputFilePacketDescriptions ? mySettings->inputFilePacketDescriptions : nil));
if (error || !ioOutputDataPackets)
{
// fprintf(stderr, "err: %ld, packets: %ld\n", err, ioOutputDataPackets);
break; // this is our termination condition
}
// write the converted data to the output file
// KEVIN: QUESTION: 3rd arg seems like it should be a byte count, not packets. why does this work?
CheckResult (AudioFileWritePackets(mySettings->outputFile,
FALSE,
ioOutputDataPackets,
NULL,
outputFilePacketPosition / mySettings->outputFormat.mBytesPerPacket,
&ioOutputDataPackets,
convertedData.mBuffers[0].mData),
"Couldn't write packets to file");
// advance the output file write location
outputFilePacketPosition += (ioOutputDataPackets * mySettings->outputFormat.mBytesPerPacket);
}
AudioConverterDispose(audioConverter);
}
