I would like to use the Core Audio extended audio file services framework to read a mp3 file, process it as a PCM, then write the modified file back as a mp3 file. I am able to convert the mp3 file to PCM, but am NOT able to write the PCM file back as a mp3.
I have followed and analyzed the Apple ExtAudioFileConvertTest sample and also cannot get that to work. The failure point is when I set the client format for the output file(set to a canonical PCM type). This fails with error "fmt?" if the output target type is set to mp3.
Is it possible to do mp3 -> PCM -> mp3 on the iPhone? If I remove the failing line, setting the kExtAudioFileProperty_ClientDataFormat for the output file, the code fails with "pkd?" when I try to write to the output file later. So basically I have 2 errors:
1) "fmt?" when trying to set kExtAudioFileProperty_ClientDataFormat for the output file
2) "pkd?" when trying to write to the output file
Here is the code to set up the files:
NSURL *fileUrl = [NSURL fileURLWithPath:sourceFilePath];
OSStatus error = noErr;
//
// Open the file
//
error = ExtAudioFileOpenURL((CFURLRef)fileUrl, &sourceFile);
if(error){
NSLog(#"AudioClip: Error opening file at %#. Error code %d", sourceFilePath, error);
return NO;
}
//
// Store the number of frames in the file
//
SInt64 numberOfFrames = 0;
UInt32 propSize = sizeof(SInt64);
error = ExtAudioFileGetProperty(sourceFile, kExtAudioFileProperty_FileLengthFrames, &propSize, &numberOfFrames);
if(error){
NSLog(#"AudioClip: Error retreiving number of frames: %d", error);
[self closeAudioFile];
return NO;
}
frameCount = numberOfFrames;
//
// Get the source file format info
//
propSize = sizeof(sourceFileFormat);
memset(&sourceFileFormat, 0, sizeof(AudioStreamBasicDescription));
error = ExtAudioFileGetProperty(sourceFile, kExtAudioFileProperty_FileDataFormat, &propSize, &sourceFileFormat);
if(error){
NSLog(#"AudioClip: Error getting source audio file properties: %d", error);
[self closeAudioFile];
return NO;
}
//
// Set the format for our read. We read in PCM, clip, then write out mp3
//
memset(&readFileFormat, 0, sizeof(AudioStreamBasicDescription));
readFileFormat.mFormatID = kAudioFormatLinearPCM;
readFileFormat.mSampleRate = 44100;
readFileFormat.mFormatFlags = kAudioFormatFlagsCanonical | kAudioFormatFlagIsNonInterleaved;
readFileFormat.mChannelsPerFrame = 1;
readFileFormat.mBitsPerChannel = 8 * sizeof(AudioSampleType);
readFileFormat.mFramesPerPacket = 1;
readFileFormat.mBytesPerFrame = sizeof(AudioSampleType);
readFileFormat.mBytesPerPacket = sizeof(AudioSampleType);
readFileFormat.mReserved = 0;
propSize = sizeof(readFileFormat);
error = ExtAudioFileSetProperty(sourceFile, kExtAudioFileProperty_ClientDataFormat, propSize, &readFileFormat);
if(error){
NSLog(#"AudioClip: Error setting read format: %d", error);
[self closeAudioFile];
return NO;
}
//
// Set the format for the output file that we will write
//
propSize = sizeof(targetFileFormat);
memset(&targetFileFormat, 0, sizeof(AudioStreamBasicDescription));
targetFileFormat.mFormatID = kAudioFormatMPEGLayer3;
targetFileFormat.mChannelsPerFrame = 1;
//
// Let the API fill in the rest
//
error = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &propSize, &targetFileFormat);
if(error){
NSLog(#"AudioClip: Error getting target file format info: %d", error);
[self closeAudioFile];
return NO;
}
//
// Create our target file
//
NSURL *writeURL = [NSURL fileURLWithPath:targetFilePath];
error = ExtAudioFileCreateWithURL( (CFURLRef)writeURL, kAudioFileMP3Type,
&targetFileFormat, NULL,
kAudioFileFlags_EraseFile,
&targetFile);
if(error){
NSLog(#"AudioClip: Error opening target file for writing: %d", error);
[self closeAudioFile];
return NO;
}
//
// Set the client format for the output file the same as our client format for the input file
//
propSize = sizeof(readFileFormat);
error = ExtAudioFileSetProperty(targetFile, kExtAudioFileProperty_ClientDataFormat, propSize, &readFileFormat);
if(error){
NSLog(#"AudioClip: Error, cannot set client format for output file: %d", error);
[self closeAudioFile];
return NO;
}
And the code to read/write:
NSInteger framesToRead = finalFrameNumber - startFrameNumber;
while(framesToRead > 0){
//
// Read frames into our data
//
short *data = (short *)malloc(framesToRead * sizeof(short));
if(!data){
NSLog(#"AudioPlayer: Cannot init memory for read buffer");
[self notifyDelegateFailure];
[self closeAudioFile];
return;
}
AudioBufferList bufferList;
OSStatus error = noErr;
UInt32 loadedPackets = framesToRead;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0].mNumberChannels = 1;
bufferList.mBuffers[0].mData = data;
bufferList.mBuffers[0].mDataByteSize = (framesToRead * sizeof(short));
NSLog(#"AudioClip: Before read nNumberBuffers = %d, mNumberChannels = %d, mData = %p, mDataByteSize = %d",
bufferList.mNumberBuffers, bufferList.mBuffers[0].mNumberChannels, bufferList.mBuffers[0].mData,
bufferList.mBuffers[0].mDataByteSize);
error = ExtAudioFileRead(sourceFile, &loadedPackets, &bufferList);
if(error){
NSLog(#"AudioClip: Error %d from ExtAudioFileRead", error);
[self notifyDelegateFailure];
[self closeAudioFile];
return;
}
//
// Now write the data to our file which will convert it into a mp3 file
//
NSLog(#"AudioClip: After read nNumberBuffers = %d, mNumberChannels = %d, mData = %p, mDataByteSize = %d",
bufferList.mNumberBuffers, bufferList.mBuffers[0].mNumberChannels, bufferList.mBuffers[0].mData,
bufferList.mBuffers[0].mDataByteSize);
error = ExtAudioFileWrite(targetFile, loadedPackets, &bufferList);
if(error){
NSLog(#"AudioClip: Error %d from ExtAudioFileWrite", error);
[self notifyDelegateFailure];
[self closeAudioFile];
return;
}
framesToRead -= loadedPackets;
}
Apple doesn't supply an MP3 encoder- only a decoder. The source document is a bit outdated, but AFAIK it is still current: http://developer.apple.com/library/ios/#documentation/MusicAudio/Conceptual/CoreAudioOverview/SupportedAudioFormatsMacOSX/SupportedAudioFormatsMacOSX.html%23//apple_ref/doc/uid/TP40003577-CH7-SW1
I think your best bet might be to use AAC.
Related
everyone , I have a problem about the API-- alSourceUnqueueBuffers when I use the OpenAL Libaray.
My problem as follows:
1.I play a pcm-music though streaming mechanism.
2.The application can queue up one or multiple buffer names using alSourceQueueBuffers.
when a buffer has been processed. I want to fill new audio data in my function: getSourceState . but when I use the API of OpenAL alSourceUnqueueBuffers. it returns an error
--- AL_INVALID_OPERATION . I do this as the document about the OpenAL.
so I test a way to solve this problem. I use alSourceStop(source) before the api alSourceUnqueueBuffers, an use alSourcePlay(source) after i filled new data though
alBufferData & alSourceQueueBuffers. but it is bad. because It breaks down the music.
who can help me to find this problem ?
and where i can find more information and method about openAL?
I am waiting for your help . thanks , everyone.
so my code as follows:
.h:
#interface myPlayback : NSObject
{
ALuint source;
ALuint * buffers;
ALCcontext* context;
ALCdevice* device;
unsigned long long offset;
ALenum m_format;
ALsizei m_freq;
void* data;
}
#end
.m
- (void)initOpenAL
{
ALenum error;
// Create a new OpenAL Device
// Pass NULL to specify the system’s default output device
device = alcOpenDevice(NULL);
if (device != NULL)
{
// Create a new OpenAL Context
// The new context will render to the OpenAL Device just created
context = alcCreateContext(device, 0);
if (context != NULL)
{
// Make the new context the Current OpenAL Context
alcMakeContextCurrent(context);
// Create some OpenAL Buffer Objects
buffers = (ALuint*)malloc(sizeof(ALuint) * 5);
alGenBuffers(5, buffers);
if((error = alGetError()) != AL_NO_ERROR) {
NSLog(#"Error Generating Buffers: %x", error);
exit(1);
}
// Create some OpenAL Source Objects
alGenSources(1, &source);
if(alGetError() != AL_NO_ERROR)
{
NSLog(#"Error generating sources! %x\n", error);
exit(1);
}
}
}
// clear any errors
alGetError();
[self initBuffer];
[self initSource];
}
- (void) initBuffer
{
ALenum error = AL_NO_ERROR;
ALenum format;
ALsizei size;
ALsizei freq;
NSBundle* bundle = [NSBundle mainBundle];
// get some audio data from a wave file
CFURLRef fileURL = (CFURLRef)[[NSURL fileURLWithPath:[bundle pathForResource:#"4" ofType:#"caf"]] retain];
if (fileURL)
{
data = MyGetOpenALAudioData(fileURL, &size, &format, &freq);
CFRelease(fileURL);
m_freq = freq;
m_format = format;
if((error = alGetError()) != AL_NO_ERROR) {
NSLog(#"error loading sound: %x\n", error);
exit(1);
}
alBufferData(buffers[0], format, data, READ_SIZE , freq);
offset += READ_SIZE;
alBufferData(buffers[1], format, data + offset, READ_SIZE, freq);
offset += READ_SIZE;
alBufferData(buffers[2], format, data + offset, READ_SIZE, freq);
offset += READ_SIZE;
alBufferData(buffers[3], format, data + offset, READ_SIZE, freq);
offset += READ_SIZE;
alBufferData(buffers[4], format, data + offset, READ_SIZE, freq);
offset += READ_SIZE;
if((error = alGetError()) != AL_NO_ERROR) {
NSLog(#"error attaching audio to buffer: %x\n", error);
}
}
else
NSLog(#"Could not find file!\n");
}
- (void) initSource
{
ALenum error = AL_NO_ERROR;
alGetError(); // Clear the error
// Turn Looping ON
alSourcei(source, AL_LOOPING, AL_TRUE);
// Set Source Position
float sourcePosAL[] = {sourcePos.x, kDefaultDistance, sourcePos.y};
alSourcefv(source, AL_POSITION, sourcePosAL);
// Set Source Reference Distance
alSourcef(source, AL_REFERENCE_DISTANCE, 50.0f);
alSourceQueueBuffers(source, 5, buffers);
if((error = alGetError()) != AL_NO_ERROR) {
NSLog(#"Error attaching buffer to source: %x\n", error);
exit(1);
}
}
- (void)startSound
{
ALenum error;
NSLog(#"Start!\n");
// Begin playing our source file
alSourcePlay(source);
if((error = alGetError()) != AL_NO_ERROR) {
NSLog(#"error starting source: %x\n", error);
} else {
// Mark our state as playing (the view looks at this)
self.isPlaying = YES;
}
while (1) {
[self getSourceState];
}
}
-(void)getSourceState
{
int queued;
int processed;
int state;
alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
alGetSourcei(source, AL_BUFFERS_PROCESSED, &processed);
alGetSourcei(source, AL_SOURCE_STATE, &state);
NSLog(#"%d", queued);
NSLog(#"%d", processed);
NSLog(#"===================================");
while (processed > 0) {
for (int i = 0; i < processed; ++i) {
ALuint buf;
alGetError();
// alSourceStop(source);
ALenum y = alGetError();
NSLog(#"%d", y);
alSourceUnqueueBuffers(source, 1, &buf);
ALenum i = alGetError();
NSLog(#"%d", i);
processed --;
alBufferData(buf, m_format, data + offset, READ_SIZE, m_freq);
ALenum j = alGetError();
NSLog(#"%d", j);
alSourceQueueBuffers(source, 1, &buf);
ALenum k = alGetError();
NSLog(#"%d", k);
offset += READ_SIZE;
// alSourcePlay(source);
}
}
// [self getSourceState];
}
I found the reason about the problem.
the reason I turn Looping ON : alSourcei(source, AL_LOOPING, AL_TRUE);
if you set this , when the source processed a buffer, you want to fill new data or delete the buffer from the source. you will get the error.
i am developing an audio player. I successfully get other information of the mp3 file. but is unable to get the album art of the mp3 file. using this code i get the mp3 file info.
- (NSDictionary *)songID3Tags
{
AudioFileID fileID = nil;
OSStatus error = noErr;
error = AudioFileOpenURL((CFURLRef)self.filePath, kAudioFileReadPermission, 0, &fileID);
if (error != noErr) {
NSLog(#"AudioFileOpenURL failed");
}
UInt32 id3DataSize = 0;
char *rawID3Tag = NULL;
error = AudioFileGetPropertyInfo(fileID, kAudioFilePropertyID3Tag, &id3DataSize, NULL);
if (error != noErr)
NSLog(#"AudioFileGetPropertyInfo failed for ID3 tag");
rawID3Tag = (char *)malloc(id3DataSize);
if (rawID3Tag == NULL)
NSLog(#"could not allocate %lu bytes of memory for ID3 tag", id3DataSize);
error = AudioFileGetProperty(fileID, kAudioFilePropertyID3Tag, &id3DataSize, rawID3Tag);
if( error != noErr )
NSLog(#"AudioFileGetPropertyID3Tag failed");
UInt32 id3TagSize = 0;
UInt32 id3TagSizeLength = 0;
error = AudioFormatGetProperty(kAudioFormatProperty_ID3TagSize, id3DataSize, rawID3Tag, &id3TagSizeLength, &id3TagSize);
if (error != noErr) {
NSLog( #"AudioFormatGetProperty_ID3TagSize failed" );
switch(error) {
case kAudioFormatUnspecifiedError:
NSLog( #"Error: audio format unspecified error" );
break;
case kAudioFormatUnsupportedPropertyError:
NSLog( #"Error: audio format unsupported property error" );
break;
case kAudioFormatBadPropertySizeError:
NSLog( #"Error: audio format bad property size error" );
break;
case kAudioFormatBadSpecifierSizeError:
NSLog( #"Error: audio format bad specifier size error" );
break;
case kAudioFormatUnsupportedDataFormatError:
NSLog( #"Error: audio format unsupported data format error" );
break;
case kAudioFormatUnknownFormatError:
NSLog( #"Error: audio format unknown format error" );
break;
default:
NSLog( #"Error: unknown audio format error" );
break;
}
}
CFDictionaryRef piDict = nil;
UInt32 piDataSize = sizeof(piDict);
error = AudioFileGetProperty(fileID, kAudioFilePropertyInfoDictionary, &piDataSize, &piDict);
if (error != noErr)
NSLog(#"AudioFileGetProperty failed for property info dictionary");
free(rawID3Tag);
return (NSDictionary*)piDict;
}
I know through kAudioFilePropertyAlbumArtwork i can get the album art of the mp3 file, but I do not know how to get it.
- (NSArray *)artworksForFileAtPath:(NSString *)path {
NSMutableArray *artworkImages = [NSMutableArray array];
NSURL *u = [NSURL fileURLWithPath:path];
AVURLAsset *a = [AVURLAsset URLAssetWithURL:u options:nil];
NSArray *artworks = [AVMetadataItem metadataItemsFromArray:a.commonMetadata withKey:AVMetadataCommonKeyArtwork keySpace:AVMetadataKeySpaceCommon];
for (AVMetadataItem *i in artworks)
{
NSString *keySpace = i.keySpace;
UIImage *im = nil;
if ([keySpace isEqualToString:AVMetadataKeySpaceID3])
{
NSDictionary *d = [i.value copyWithZone:nil];
im = [UIImage imageWithData:[d objectForKey:#"data"]];
}
else if ([keySpace isEqualToString:AVMetadataKeySpaceiTunes])
im = [UIImage imageWithData:[i.value copyWithZone:nil]];
if (im)
[artworkImages addObject:im];
}
NSLog(#"array description is %#", [artworkImages description]);
return artworkImages; }
The Above code return the album_art of the mp3 file. Where path is the audio file path.
I am converting my recorded audio which is in .m4a format to .caf format. The settings of the recorded audio is as given below:
/* Record settings for recording the audio*/
recordSetting = [[NSDictionary alloc] initWithObjectsAndKeys:[NSNumber numberWithInt:kAudioFormatMPEG4AAC],AVFormatIDKey,
[NSNumber numberWithInt:44100.0],AVSampleRateKey,
[NSNumber numberWithInt: 2],AVNumberOfChannelsKey,
[NSNumber numberWithInt:16],AVLinearPCMBitDepthKey,
[NSNumber numberWithBool:NO],AVLinearPCMIsBigEndianKey,
[NSNumber numberWithBool:NO],AVLinearPCMIsFloatKey,
nil];
I convert the audio to .caf using this function:
-(NSString *)handleConvertToPCM:(NSURL *)convertUrl
{
[self performSelectorOnMainThread:#selector(showActivity) withObject:nil waitUntilDone:NO];
DEBUG_LOG(#"DEBUGGING");
DEBUG_LOG(#"handleConvertToPCM");
// open an ExtAudioFile
NSLog (#"opening %#", convertUrl);
ExtAudioFileRef inputFile;
CheckResult (ExtAudioFileOpenURL((CFURLRef)convertUrl, &inputFile),
"ExtAudioFileOpenURL failed");
// prepare to convert to a plain ol' PCM format
AudioStreamBasicDescription requiredPCMFormat;
requiredPCMFormat.mSampleRate = 44100; // todo: or use source rate?
requiredPCMFormat.mFormatID = kAudioFormatLinearPCM ;
requiredPCMFormat.mFormatFlags = kAudioFormatFlagsCanonical;
requiredPCMFormat.mChannelsPerFrame = 2;
requiredPCMFormat.mFramesPerPacket = 1;
requiredPCMFormat.mBitsPerChannel = 16;
requiredPCMFormat.mBytesPerPacket = 4;
requiredPCMFormat.mBytesPerFrame = 4;
CheckResult (ExtAudioFileSetProperty(inputFile, kExtAudioFileProperty_ClientDataFormat,
sizeof (requiredPCMFormat), &requiredPCMFormat),
"ExtAudioFileSetProperty failed");
// allocate a big buffer. size can be arbitrary for ExtAudioFile.
UInt32 outputBufferSize = 0x10000;
void* ioBuf = malloc (outputBufferSize);
UInt32 sizePerPacket = requiredPCMFormat.mBytesPerPacket;
UInt32 packetsPerBuffer = outputBufferSize / sizePerPacket;
// set up output file
self.outputPath = [NSString stringWithFormat:#"%#/export-pcm.caf",DOCUMENTS_FOLDER];
self.outputURL = [NSURL fileURLWithPath:self.outputPath];
DEBUG_LOG(#"creating output file %#", self.outputURL);
AudioFileID outputFile;
CheckResult(AudioFileCreateWithURL((CFURLRef)outputURL,
kAudioFileCAFType,
&requiredPCMFormat,
kAudioFileFlags_EraseFile,
&outputFile),
"AudioFileCreateWithURL failed");
// start convertin'
UInt32 outputFilePacketPosition = 0; //in bytes
while (true)
{
// wrap the destination buffer in an AudioBufferList
AudioBufferList convertedData;
convertedData.mNumberBuffers = 1;
convertedData.mBuffers[0].mNumberChannels = requiredPCMFormat.mChannelsPerFrame;
convertedData.mBuffers[0].mDataByteSize = outputBufferSize;
convertedData.mBuffers[0].mData = ioBuf;
UInt32 frameCount = packetsPerBuffer;
// read from the extaudiofile
CheckResult (ExtAudioFileRead(inputFile,
&frameCount,
&convertedData),
"Couldn't read from input file");
if (frameCount == 0)
{
printf ("done reading from file");
break;
}
// write the converted data to the output file
CheckResult (AudioFileWritePackets(outputFile,
false,
frameCount,
NULL,
outputFilePacketPosition / requiredPCMFormat.mBytesPerPacket,
&frameCount,
convertedData.mBuffers[0].mData),
"Couldn't write packets to file");
DEBUG_LOG(#"Converted %ld bytes", outputFilePacketPosition);
// advance the output file write location
outputFilePacketPosition += (frameCount * requiredPCMFormat.mBytesPerPacket);
}
// clean up
ExtAudioFileDispose(inputFile);
AudioFileClose(outputFile);
return(self.outputPath);
}
My problem is that the size of the converted file is very high compared to the file given for conversion.Is there anyway to decrease the size by changing the conversion settings.
I tried compressing the file obtained , but it takes much time to compress.So I would like to get a way to decrease size along with conversion.
Decompressing a highly compressed audio file almost always results in a much larger result file, unless you re-compress using an even lossier compression format.
I need to convert a WAVE file into an AAC encoded M4A file on iOS. I'm aware that AAC encoding is not supported on older devices or in the simulator. I'm testing that before I run the code. But I still can't get it to work.
I looked into Apple's very own iPhoneExtAudioFileConvertTest example and I thought I followed it exactly, but still no luck!
Currently, I get a -50 (= error in user parameter list) while trying to set the client format on the destination file. On the source file, it works.
Below is my code. Any help is very much appreciated, thanks!
UInt32 size;
// Open a source audio file.
ExtAudioFileRef sourceAudioFile;
ExtAudioFileOpenURL( (CFURLRef)sourceURL, &sourceAudioFile );
// Get the source data format
AudioStreamBasicDescription sourceFormat;
size = sizeof( sourceFormat );
result = ExtAudioFileGetProperty( sourceAudioFile, kExtAudioFileProperty_FileDataFormat, &size, &sourceFormat );
// Define the output format (AAC).
AudioStreamBasicDescription outputFormat;
outputFormat.mFormatID = kAudioFormatMPEG4AAC;
outputFormat.mSampleRate = 44100;
outputFormat.mChannelsPerFrame = 2;
// Use AudioFormat API to fill out the rest of the description.
size = sizeof( outputFormat );
AudioFormatGetProperty( kAudioFormatProperty_FormatInfo, 0, NULL, &size, &outputFormat);
// Make a destination audio file with this output format.
ExtAudioFileRef destAudioFile;
ExtAudioFileCreateWithURL( (CFURLRef)destURL, kAudioFileM4AType, &outputFormat, NULL, kAudioFileFlags_EraseFile, &destAudioFile );
// Create canonical PCM client format.
AudioStreamBasicDescription clientFormat;
clientFormat.mSampleRate = sourceFormat.mSampleRate;
clientFormat.mFormatID = kAudioFormatLinearPCM;
clientFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;
clientFormat.mChannelsPerFrame = 2;
clientFormat.mBitsPerChannel = 16;
clientFormat.mBytesPerFrame = 4;
clientFormat.mBytesPerPacket = 4;
clientFormat.mFramesPerPacket = 1;
// Set the client format in source and destination file.
size = sizeof( clientFormat );
ExtAudioFileSetProperty( sourceAudioFile, kExtAudioFileProperty_ClientDataFormat, size, &clientFormat );
size = sizeof( clientFormat );
ExtAudioFileSetProperty( destAudioFile, kExtAudioFileProperty_ClientDataFormat, size, &clientFormat );
// Make a buffer
int bufferSizeInFrames = 8000;
int bufferSize = ( bufferSizeInFrames * sourceFormat.mBytesPerFrame );
UInt8 * buffer = (UInt8 *)malloc( bufferSize );
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0].mNumberChannels = clientFormat.mChannelsPerFrame;
bufferList.mBuffers[0].mData = buffer;
bufferList.mBuffers[0].mDataByteSize = ( bufferSize );
while( TRUE )
{
// Try to fill the buffer to capacity.
UInt32 framesRead = bufferSizeInFrames;
ExtAudioFileRead( sourceAudioFile, &framesRead, &bufferList );
// 0 frames read means EOF.
if( framesRead == 0 )
break;
// Write.
ExtAudioFileWrite( destAudioFile, framesRead, &bufferList );
}
free( buffer );
// Close the files.
ExtAudioFileDispose( sourceAudioFile );
ExtAudioFileDispose( destAudioFile );
Answered my own question: I had to pass this problem to my colleague and he got it to work! I never had the chance to analyze my original problem but I thought, I'd post it here for the sake of completeness. The following method is called from within an NSThread. Parameters are set via the 'threadDictionary' and he created a custom delegate to transmit progress feedback (sorry, SO doesn't understand the formatting properly, the following is supposed to be one block of method implementation):
- (void)encodeToAAC
{
RXAudioEncoderStatusType encoderStatus;
OSStatus result = noErr;
BOOL success = NO;
BOOL cancelled = NO;
UInt32 size;
ExtAudioFileRef sourceAudioFile,destAudioFile;
AudioStreamBasicDescription sourceFormat,outputFormat, clientFormat;
SInt64 totalFrames;
unsigned long long encodedBytes, totalBytes;
int bufferSizeInFrames, bufferSize;
UInt8 * buffer;
AudioBufferList bufferList;
NSAutoreleasePool * pool = [[NSAutoreleasePool alloc] init];
NSFileManager * fileManager = [[[NSFileManager alloc] init] autorelease];
NSMutableDictionary * threadDict = [[NSThread currentThread] threadDictionary];
NSObject<RXAudioEncodingDelegate> * delegate = (NSObject<RXAudioEncodingDelegate> *)[threadDict objectForKey:#"Delegate"];
NSString *sourcePath = (NSString *)[threadDict objectForKey:#"SourcePath"];
NSString *destPath = (NSString *)[threadDict objectForKey:#"DestinationPath"];
NSURL * sourceURL = [NSURL fileURLWithPath:sourcePath];
NSURL * destURL = [NSURL fileURLWithPath:destPath];
// Open a source audio file.
result = ExtAudioFileOpenURL( (CFURLRef)sourceURL, &sourceAudioFile );
if( result != noErr )
{
DLog( #"Error in ExtAudioFileOpenURL: %ld", result );
goto bailout;
}
// Get the source data format
size = sizeof( sourceFormat );
result = ExtAudioFileGetProperty( sourceAudioFile, kExtAudioFileProperty_FileDataFormat, &size, &sourceFormat );
if( result != noErr )
{
DLog( #"Error in ExtAudioFileGetProperty: %ld", result );
goto bailout;
}
// Define the output format (AAC).
memset(&outputFormat, 0, sizeof(outputFormat));
outputFormat.mFormatID = kAudioFormatMPEG4AAC;
outputFormat.mSampleRate = 44100;
outputFormat.mFormatFlags = kMPEG4Object_AAC_Main;
outputFormat.mChannelsPerFrame = 2;
outputFormat.mBitsPerChannel = 0;
outputFormat.mBytesPerFrame = 0;
outputFormat.mBytesPerPacket = 0;
outputFormat.mFramesPerPacket = 1024;
// Use AudioFormat API to fill out the rest of the description.
//size = sizeof( outputFormat );
//AudioFormatGetProperty( kAudioFormatProperty_FormatInfo, 0, NULL, &size, &outputFormat);
// Make a destination audio file with this output format.
result = ExtAudioFileCreateWithURL( (CFURLRef)destURL, kAudioFileM4AType, &outputFormat, NULL, kAudioFileFlags_EraseFile, &destAudioFile );
if( result != noErr )
{
DLog( #"Error creating destination file: %ld", result );
goto bailout;
}
// Create the canonical PCM client format.
memset(&clientFormat, 0, sizeof(clientFormat));
clientFormat.mSampleRate = sourceFormat.mSampleRate;
clientFormat.mFormatID = kAudioFormatLinearPCM;
clientFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked; //kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;
clientFormat.mChannelsPerFrame = 2;
clientFormat.mBitsPerChannel = 16;
clientFormat.mBytesPerFrame = 4;
clientFormat.mBytesPerPacket = 4;
clientFormat.mFramesPerPacket = 1;
// Set the client format in source and destination file.
size = sizeof( clientFormat );
result = ExtAudioFileSetProperty( sourceAudioFile, kExtAudioFileProperty_ClientDataFormat, size, &clientFormat );
if( result != noErr )
{
DLog( #"Error while setting client format in source file: %ld", result );
goto bailout;
}
size = sizeof( clientFormat );
result = ExtAudioFileSetProperty( destAudioFile, kExtAudioFileProperty_ClientDataFormat, size, &clientFormat );
if( result != noErr )
{
DLog( #"Error while setting client format in destination file: %ld", result );
goto bailout;
}
// Make a buffer
bufferSizeInFrames = 8000;
bufferSize = ( bufferSizeInFrames * sourceFormat.mBytesPerFrame );
buffer = (UInt8 *)malloc( bufferSize );
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0].mNumberChannels = clientFormat.mChannelsPerFrame;
bufferList.mBuffers[0].mData = buffer;
bufferList.mBuffers[0].mDataByteSize = ( bufferSize );
// Obtain total number of audio frames to encode
size = sizeof( totalFrames );
result = ExtAudioFileGetProperty( sourceAudioFile, kExtAudioFileProperty_FileLengthFrames, &size, &totalFrames );
if( result != noErr )
{
DLog( #"Error in ExtAudioFileGetProperty, could not get kExtAudioFileProperty_FileLengthFrames from sourceFile: %ld", result );
goto bailout;
}
encodedBytes = 0;
totalBytes = totalFrames * sourceFormat.mBytesPerFrame;
[threadDict setValue:[NSValue value:&totalBytes withObjCType:#encode(unsigned long long)] forKey:#"TotalBytes"];
if (delegate != nil)
[self performSelectorOnMainThread:#selector(didStartEncoding) withObject:nil waitUntilDone:NO];
while( TRUE )
{
// Try to fill the buffer to capacity.
UInt32 framesRead = bufferSizeInFrames;
result = ExtAudioFileRead( sourceAudioFile, &framesRead, &bufferList );
if( result != noErr )
{
DLog( #"Error in ExtAudioFileRead: %ld", result );
success = NO;
break;
}
// 0 frames read means EOF.
if( framesRead == 0 ) {
success = YES;
break;
}
// Write.
result = ExtAudioFileWrite( destAudioFile, framesRead, &bufferList );
if( result != noErr )
{
DLog( #"Error in ExtAudioFileWrite: %ld", result );
success = NO;
break;
}
encodedBytes += framesRead * sourceFormat.mBytesPerFrame;
if (delegate != nil)
[self performSelectorOnMainThread:#selector(didEncodeBytes:) withObject:[NSValue value:&encodedBytes withObjCType:#encode(unsigned long long)] waitUntilDone:NO];
if ([[NSThread currentThread] isCancelled]) {
cancelled = YES;
DLog( #"Encoding was cancelled." );
success = NO;
break;
}
}
free( buffer );
// Close the files.
ExtAudioFileDispose( sourceAudioFile );
ExtAudioFileDispose( destAudioFile );
bailout:
encoderStatus.result = result;
[threadDict setValue:[NSValue value:&encoderStatus withObjCType:#encode(RXAudioEncoderStatusType)] forKey:#"EncodingError"];
// Report to the delegate if one exists
if (delegate != nil)
if (success)
[self performSelectorOnMainThread:#selector(didEncodeFile) withObject:nil waitUntilDone:YES];
else if (cancelled)
[self performSelectorOnMainThread:#selector(encodingCancelled) withObject:nil waitUntilDone:YES];
else
[self performSelectorOnMainThread:#selector(failedToEncodeFile) withObject:nil waitUntilDone:YES];
// Clear the partially encoded file if encoding failed or is cancelled midway
if ((cancelled || !success) && [fileManager fileExistsAtPath:destPath])
[fileManager removeItemAtURL:destURL error:NULL];
[threadDict setValue:[NSNumber numberWithBool:NO] forKey:#"isEncoding"];
[pool release];
}
Are you sure the sample rates match? Can you print the values for clientFormat and outputFormat at the point you’re getting the error? Otherwise I think you might need an AudioConverter.
I tried out the code in Sebastian's answer and while it worked for uncompressed files (aif, wav, caf), it didn't for a lossy compressed file (mp3). I also had an error code of -50, but in ExtAudioFileRead rather than ExtAudioFileSetProperty. From this question I learned that this error signifies a problem with the function parameters. Turns out the buffer for reading the audio file had a size of 0 bytes, a result of this line:
int bufferSize = ( bufferSizeInFrames * sourceFormat.mBytesPerFrame );
Switching it to use the the bytes per frame from clientFormat instead (sourceFormat's value was 0) worked for me:
int bufferSize = ( bufferSizeInFrames * clientFormat.mBytesPerFrame );
This line was also in the question code, but I don't think that was the problem (but I had too much text for a comment).
I'm following a tutorial about playing sound with OpenAL. Now everything works fine except I can't make the sound looping. I believe that I've used AL_LOOPING for the source. Now it can only play once and when it finishes playing, the app will block(doesn't response to my tap on the play button). Any ideas about what's wrong with the code?
// start up openAL
// init device and context
-(void)initOpenAL
{
// Initialization
mDevice = alcOpenDevice(NULL); // select the "preferred device"
if (mDevice) {
// use the device to make a context
mContext = alcCreateContext(mDevice, NULL);
// set my context to the currently active one
alcMakeContextCurrent(mContext);
}
}
// open the audio file
// returns a big audio ID struct
-(AudioFileID)openAudioFile:(NSString*)filePath
{
AudioFileID outAFID;
// use the NSURl instead of a cfurlref cuz it is easier
NSURL * afUrl = [NSURL fileURLWithPath:filePath];
// do some platform specific stuff..
#if TARGET_OS_IPHONE
OSStatus result = AudioFileOpenURL((CFURLRef)afUrl, kAudioFileReadPermission, 0, &outAFID);
#else
OSStatus result = AudioFileOpenURL((CFURLRef)afUrl, fsRdPerm, 0, &outAFID);
#endif
if (result != 0) NSLog(#"cannot openf file: %#",filePath);
return outAFID;
}
// find the audio portion of the file
// return the size in bytes
-(UInt32)audioFileSize:(AudioFileID)fileDescriptor
{
UInt64 outDataSize = 0;
UInt32 thePropSize = sizeof(UInt64);
OSStatus result = AudioFileGetProperty(fileDescriptor, kAudioFilePropertyAudioDataByteCount, &thePropSize, &outDataSize);
if(result != 0) NSLog(#"cannot find file size");
return (UInt32)outDataSize;
}
- (void)stopSound
{
alSourceStop(sourceID);
}
-(void)cleanUpOpenAL:(id)sender
{
// delete the sources
alDeleteSources(1, &sourceID);
// delete the buffers
alDeleteBuffers(1, &bufferID);
// destroy the context
alcDestroyContext(mContext);
// close the device
alcCloseDevice(mDevice);
}
-(IBAction)play:(id)sender
{
alSourcePlay(sourceID);
}
#pragma mark -
// Implement viewDidLoad to do additional setup after loading the view, typically from a nib.
- (void)viewDidLoad {
[super viewDidLoad];
[self initOpenAL];
// get the full path of the file
NSString* fileName = [[NSBundle mainBundle] pathForResource:#"sound" ofType:#"caf"];
// first, open the file
AudioFileID fileID = [self openAudioFile:fileName];
// find out how big the actual audio data is
UInt32 fileSize = [self audioFileSize:fileID];
// this is where the audio data will live for the moment
unsigned char * outData = malloc(fileSize);
// this where we actually get the bytes from the file and put them
// into the data buffer
OSStatus result = noErr;
result = AudioFileReadBytes(fileID, false, 0, &fileSize, outData);
AudioFileClose(fileID); //close the file
if (result != 0) NSLog(#"cannot load effect: %#", fileName);
//NSUInteger bufferID; // buffer is defined in head file
// grab a buffer ID from openAL
alGenBuffers(1, &bufferID);
// jam the audio data into the new buffer
alBufferData(bufferID, AL_FORMAT_STEREO16, outData, fileSize, 8000);
//NSUInteger sourceID; // source is defined in head file
// grab a source ID from openAL
alGenSources(1, &sourceID);
// attach the buffer to the source
alSourcei(sourceID, AL_BUFFER, bufferID);
// set some basic source prefs
alSourcef(sourceID, AL_PITCH, 1.0f);
alSourcef(sourceID, AL_GAIN, 1.0f);
alSourcei(sourceID, AL_LOOPING, AL_TRUE);
// clean up the buffer
if (outData)
{
free(outData);
outData = NULL;
}
}
You should be able to release outData right after your alBufferData() call. It exclude it as the culprit, you can try the static extension and manage the memory yourself. It's something like:
alBufferDataStaticProcPtr alBufferDataStaticProc = (alBufferDataStaticProcPtr)alcGetProcAddress(0, (const ALCchar *)"alBufferDataStatic");
alBufferDataStaticProc(bufferID, bitChanFormat, audioData, audioDataSize, dataFormat.mSampleRate);