I am migrating from audiobus sdk 1.0 to 2.0 to support ios8. but I am facing some problem with functionality code.
What I want to achieve: my app is a receiver,I want to play received audio with ABReceiverPortReceive method.
In old sdk there is callback method in ABInputPort when audio available but in sdk 2.0 no callback method is there in ABReceiverPort
Following is my old code of sdk 1.0
_inputPort.audioInputBlock = ^(ABInputPort *inputPort, UInt32 lengthInFrames, AudioTimeStamp *nextTimestamp, ABPort *sourcePortOrNil) {
// get samples from audiobus
AudioBufferList* receiveBufferList = alloca(sizeof(AudioBufferList) + (_receiveBufferList->mNumberBuffers - 1) * sizeof(AudioBuffer));
memcpy(receiveBufferList, _receiveBufferList, sizeof(AudioBufferList) + (_receiveBufferList->mNumberBuffers - 1) * sizeof(AudioBuffer));
ABInputPortReceive(inputPort, sourcePortOrNil, receiveBufferList, &lengthInFrames, nextTimestamp, NULL);
// put samples into buffer
if (lengthInFrames > 0)
{
ABMultiStreamBufferEnqueue([ABPlugin plugin].inputBuffer, sourcePortOrNil, receiveBufferList, lengthInFrames, nextTimestamp);
}
};
here I call ABInputPortReceive when calback method called and audio available but no callback method in new sdk.
SO I called the ABReceiverPortReceive method in my audio engine callback method.
- (OSStatus)render:(AudioBufferList *)buffer frames:(NSUInteger)numFrames timestamp:(const AudioTimeStamp *)timestamp flags:(AudioUnitRenderActionFlags *)ioFlags
{
// NSLog(#"%s", __PRETTY_FUNCTION__);
if (nil == self.source)
{
*ioFlags |= kAudioUnitRenderAction_OutputIsSilence;
return noErr;
}
// read previously enqueued by plugin samples from buffer into output buffer
UInt32 totalFrames = 0;
UInt32 framesFromBuffer = numFrames;
// get samples from input buffer
AudioBufferList* receiveBufferList = alloca(sizeof(AudioBufferList) + (_receiveBufferList->mNumberBuffers - 1) * sizeof(AudioBuffer));
memcpy(receiveBufferList, _receiveBufferList, sizeof(AudioBufferList) + (_receiveBufferList->mNumberBuffers - 1) * sizeof(AudioBuffer));
AudioTimeStamp bufferTimestamp = *timestamp;
ABMultiStreamBufferDequeueSingleSource([ABPlugin plugin].inputBuffer, self.source, receiveBufferList, &framesFromBuffer, &bufferTimestamp);
totalFrames += framesFromBuffer;
_framesPlayed += numFrames;
[ABPlugin plugin].framesPlayed = MAX([ABPlugin plugin].framesPlayed, _framesPlayed);
ABMultiStreamBufferMarkSourceIdle([ABPlugin plugin].inputBuffer, self.source);
// convert AB format to output stream format
OSStatus result = AudioConverterConvertComplexBuffer(_formatConverter, totalFrames, receiveBufferList, buffer);
AudioTimeStamp timestamp1 = *timestamp;
if ( ABReceiverPortIsConnected([ABPlugin plugin].inputPort) ) {
// Receive audio from Audiobus
ABReceiverPortReceive([ABPlugin plugin].inputPort, self.source, _receiveBufferList, numFrames, ×tamp1);
}
return result;
}
Problem : There is no audio recieved in this way
Please help or suggest something as early as possible.
Thanks.
It seems you are handling the multi stream here. may be you need to differentiate between the sources of sound. I can't help with the code.
did you see this website 1.x-2.x Migration Guide : http://developer.audiob.us/doc/_migration-_guide.html
Related
I am using EZAudio and trying to write some samples generated by my softaware to an AudioBuffer provided by this library https://github.com/syedhali/EZAudio as seen in the example "Playback By Manual Override".
My code looks like this...
// Completely override the output callback function
- (void)
output:(EZOutput *)output
callbackWithActionFlags:(AudioUnitRenderActionFlags *)ioActionFlags
inTimeStamp:(const AudioTimeStamp *)inTimeStamp
inBusNumber:(UInt32)inBusNumber
inNumberFrames:(UInt32)inNumberFrames
ioData:(AudioBufferList *)ioData {
//grab latest sample from sample queue
if (currentAudioPiece == nil || currentAudioPiece.duration >= currentAudioPieceIndex) {
self.currentAudioPiece = sampleQueue.dequeue;
}
AudioBuffer audioBuffer = ioData->mBuffers[0];
if (true) {
for (int i = 0; i < audioBuffer.mDataByteSize; i++) {
uint8_t rofl[2048];
arc4random_buf(&rofl, 2048);
audioBuffer.mData = rofl;
}
return;
}
//... more code that I'll debug later...'
Essentially I am unable to get a sanity check that a random bunch of memory playing back should make some noise. I think the problem is with "audioBuffer.mData = rofl;". I'm rather confused about working with memory at void*.
jn_pdx is correct, you need to copy/fill mData instead of reassigning it.
- (void)
output:(EZOutput *)output
callbackWithActionFlags:(AudioUnitRenderActionFlags *)ioActionFlags
inTimeStamp:(const AudioTimeStamp *)inTimeStamp
inBusNumber:(UInt32)inBusNumber
inNumberFrames:(UInt32)inNumberFrames
ioData:(AudioBufferList *)ioData {
//grab latest sample from sample queue
if (currentAudioPiece == nil || currentAudioPiece.duration >= currentAudioPieceIndex) {
self.currentAudioPiece = sampleQueue.dequeue;
}
AudioBuffer audioBuffer = ioData->mBuffers[0];
if (true) {
arc4random_buf(audioBuffer.mData, audioBuffer.mDataByteSize);
return;
}
So, basically I want to play some audio files (mp3 and caf mostly). But the callback never gets called. Only when I call them to prime the queue.
Here's my data struct:
struct AQPlayerState
{
CAStreamBasicDescription mDataFormat;
AudioQueueRef mQueue;
AudioQueueBufferRef mBuffers[kBufferNum];
AudioFileID mAudioFile;
UInt32 bufferByteSize;
SInt64 mCurrentPacket;
UInt32 mNumPacketsToRead;
AudioStreamPacketDescription *mPacketDescs;
bool mIsRunning;
};
Here's my callback function:
static void HandleOutputBuffer (void *aqData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer)
{
NSLog(#"HandleOutput");
AQPlayerState *pAqData = (AQPlayerState *) aqData;
if (pAqData->mIsRunning == false) return;
UInt32 numBytesReadFromFile;
UInt32 numPackets = pAqData->mNumPacketsToRead;
AudioFileReadPackets (pAqData->mAudioFile,
false,
&numBytesReadFromFile,
pAqData->mPacketDescs,
pAqData->mCurrentPacket,
&numPackets,
inBuffer->mAudioData);
if (numPackets > 0) {
inBuffer->mAudioDataByteSize = numBytesReadFromFile;
AudioQueueEnqueueBuffer (pAqData->mQueue,
inBuffer,
(pAqData->mPacketDescs ? numPackets : 0),
pAqData->mPacketDescs);
pAqData->mCurrentPacket += numPackets;
} else {
// AudioQueueStop(pAqData->mQueue, false);
// AudioQueueDispose(pAqData->mQueue, true);
// AudioFileClose (pAqData->mAudioFile);
// free(pAqData->mPacketDescs);
// free(pAqData->mFloatBuffer);
pAqData->mIsRunning = false;
}
}
And here's my method:
- (void)playFile
{
AQPlayerState aqData;
// get the source file
NSString *p = [[NSBundle mainBundle] pathForResource:#"1_Female" ofType:#"mp3"];
NSURL *url2 = [NSURL fileURLWithPath:p];
CFURLRef srcFile = (__bridge CFURLRef)url2;
OSStatus result = AudioFileOpenURL(srcFile, 0x1/*fsRdPerm*/, 0/*inFileTypeHint*/, &aqData.mAudioFile);
CFRelease (srcFile);
CheckError(result, "Error opinning sound file");
UInt32 size = sizeof(aqData.mDataFormat);
CheckError(AudioFileGetProperty(aqData.mAudioFile, kAudioFilePropertyDataFormat, &size, &aqData.mDataFormat),
"Error getting file's data format");
CheckError(AudioQueueNewOutput(&aqData.mDataFormat, HandleOutputBuffer, &aqData, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &aqData.mQueue),
"Error AudioQueueNewOutPut");
// 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 = (aqData.mDataFormat.mBytesPerPacket == 0 || aqData.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);
CheckError(AudioFileGetProperty(aqData.mAudioFile, kAudioFilePropertyPacketSizeUpperBound, &size, &maxPacketSize),
"Error getting max packet size");
// adjust buffer size to represent about a second of audio based on this format
CalculateBytesForTime(aqData.mDataFormat, maxPacketSize, 1.0/*seconds*/, &aqData.bufferByteSize, &aqData.mNumPacketsToRead);
if (isFormatVBR) {
aqData.mPacketDescs = new AudioStreamPacketDescription [aqData.mNumPacketsToRead];
} else {
aqData.mPacketDescs = NULL; // we don't provide packet descriptions for constant bit rate formats (like linear PCM)
}
printf ("Buffer Byte Size: %d, Num Packets to Read: %d\n", (int)aqData.bufferByteSize, (int)aqData.mNumPacketsToRead);
}
// if the file has a magic cookie, we should get it and set it on the AQ
size = sizeof(UInt32);
result = AudioFileGetPropertyInfo(aqData.mAudioFile, kAudioFilePropertyMagicCookieData, &size, NULL);
if (!result && size) {
char* cookie = new char [size];
CheckError(AudioFileGetProperty(aqData.mAudioFile, kAudioFilePropertyMagicCookieData, &size, cookie),
"Error getting cookie from file");
CheckError(AudioQueueSetProperty(aqData.mQueue, kAudioQueueProperty_MagicCookie, cookie, size),
"Error setting cookie to file");
delete[] cookie;
}
aqData.mCurrentPacket = 0;
for (int i = 0; i < kBufferNum; ++i) {
CheckError(AudioQueueAllocateBuffer (aqData.mQueue,
aqData.bufferByteSize,
&aqData.mBuffers[i]),
"Error AudioQueueAllocateBuffer");
HandleOutputBuffer (&aqData,
aqData.mQueue,
aqData.mBuffers[i]);
}
// set queue's gain
Float32 gain = 1.0;
CheckError(AudioQueueSetParameter (aqData.mQueue,
kAudioQueueParam_Volume,
gain),
"Error AudioQueueSetParameter");
aqData.mIsRunning = true;
CheckError(AudioQueueStart(aqData.mQueue,
NULL),
"Error AudioQueueStart");
}
And the output when I press play:
Buffer Byte Size: 40310, Num Packets to Read: 38
HandleOutput start
HandleOutput start
HandleOutput start
I tryed replacing CFRunLoopGetCurrent() with CFRunLoopGetMain() and CFRunLoopCommonModes with CFRunLoopDefaultMode, but nothing.
Shouldn't the primed buffers start playing right away I start the queue?
When I start the queue, no callbacks are bang fired.
What am I doing wrong? Thanks for any ideas
What you are basically trying to do here is a basic example of audio playback using Audio Queues. Without looking at your code in detail to see what's missing (that could take a while) i'd rather recommend to you to follow the steps in this basic sample code that does exactly what you're doing (without the extras that aren't really relevant.. for example why are you trying to add audio gain?)
Somewhere else you were trying to play audio using audio units. Audio units are more complex than basic audio queue playback, and I wouldn't attempt them before being very comfortable with audio queues. But you can look at this example project for a basic example of audio queues.
In general when it comes to Core Audio programming in iOS, it's best you take your time with the basic examples and build your way up.. the problem with a lot of tutorials online is that they add extra stuff and often mix it with obj-c code.. when Core Audio is purely C code (ie the extra stuff won't add anything to the learning process). I strongly recommend you go over the book Learning Core Audio if you haven't already. All the sample code is available online, but you can also clone it from this repo for convenience. That's how I learned core audio. It takes time :)
I have inBuffer->mAudioData converted into NSData coming via Network. I do this using the Audio Queue Callback.
How do I convert this NSData so that I can either create a .caf sound file or directly give output to the speaker on the other side ?
Thanks for help.
Edit 1:
Below is the code I have used on the sender side to send Data on a wifi network:
void AudioInputCallback(void * inUserData,
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp * inStartTime,
UInt32 inNumberPacketDescriptions,
const AudioStreamPacketDescription * inPacketDescs)
{
RecordState * recordState = (RecordState*)inUserData;
if (!recordState->recording)
{
printf("Not recording, returning\n");
}
// if (inNumberPacketDescriptions == 0 && recordState->dataFormat.mBytesPerPacket != 0)
// {
// inNumberPacketDescriptions = inBuffer->mAudioDataByteSize / recordState->dataFormat.mBytesPerPacket;
// }
printf("Writing buffer %lld\n", recordState->currentPacket);
OSStatus status = AudioFileWritePackets(recordState->audioFile,
false,
inBuffer->mAudioDataByteSize,
inPacketDescs,
recordState->currentPacket,
&inNumberPacketDescriptions,
inBuffer->mAudioData);
NSLog(#"DATA = %#",[NSData dataWithBytes:inBuffer->mAudioData length:inBuffer->mAudioDataByteSize*NUM_BUFFERS]);
[[NSNotificationCenter defaultCenter] postNotificationName:#"Recording" object:[NSData dataWithBytes:inBuffer->mAudioData length:inBuffer->mAudioDataByteSize*NUM_BUFFERS]];
if (status == 0)
{
recordState->currentPacket += inNumberPacketDescriptions;
}
AudioQueueEnqueueBuffer(recordState->queue, inBuffer, 0, NULL);
}
Here I have casted inBuffer->mAudioData to NSData and then I have sent to the outputStream.
On the other end that is the receivers side I have used the below code:
-(void)audioMessageData:(NSData *)audioData fromUser:(NSString *)userName {
NSLog(#"DATA = %#",audioData);
}
Whenever I get data the above function gets called and I get NSData which I have sent from the sender iPhone. Now I want to play whatever audioData I am receiving.
Thanks for the help.
The easiest way is to configure an Audio Queue or RemoteIO Audio Unit to play data of the same format (raw linear PCM?) and start pulling samples from a lock free circular buffer (playing silence for underflow). Then just copy your received data out of the NSData network packets into the circular buffer.
I've successfully recorded audio from the microphone into an audio file using Audio Units with the help of openframeworks and this website http://atastypixel.com/blog/using-remoteio-audio-unit.
I want to be able to stream the file back to audio units and play the audio. According to Play an audio file using RemoteIO and Audio Unit I can use ExtAudioFileOpenURL and ExtAudioFileRead. However, how do I play audio data in my buffer?
This is what I currently have:
static OSStatus setupAudioFileRead() {
//construct the file destination URL
CFURLRef destinationURL = audioSystemFileURL();
OSStatus status = ExtAudioFileOpenURL(destinationURL, &audioFileRef);
CFRelease(destinationURL);
if (checkStatus(status)) { ofLog(OF_LOG_ERROR, "ofxiPhoneSoundStream: Couldn't open file to read"); return status; }
while( TRUE ) {
// Try to fill the buffer to capacity.
UInt32 framesRead = 8000;
status = ExtAudioFileRead( audioFileRef, &framesRead, &inputBufferList );
// error check
if( checkStatus(status) ) { break; }
// 0 frames read means EOF.
if( framesRead == 0 ) { break; }
//play audio???
}
return noErr;
}
From this author: http://atastypixel.com/blog/using-remoteio-audio-unit/, if you scroll down to the PLAYBACK section, try something like this:
static OSStatus playbackCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
// Notes: ioData contains buffers (may be more than one!)
// Fill them up as much as you can. Remember to set the size value in each buffer to match how
// much data is in the buffer.
for (int i=0; i < ioData->mNumberBuffers; i++)
{
AudioBuffer buffer = ioData->mBuffers[i];
// copy from your whatever buffer data to output buffer
UInt32 size = min(buffer.mDataByteSize, your buffer.size);
memcpy(buffer.mData, your buffer, size);
buffer.mDataByteSize = size; // indicate how much data we wrote in the buffer
// To test if your Audio Unit setup is working - comment out the three
// lines above and uncomment the for loop below to hear random noise
/*
UInt16 *frameBuffer = buffer.mData;
for (int j = 0; j < inNumberFrames; j++) {
frameBuffer[j] = rand();
}
*/
}
return noErr;
}
If you are only looking for recording from MIC to a file and play it back, the Apple's Speakhere sample is probably much more ready to use.
Basically,
1. Create a RemoteIO unit (See references about how to create RemoteIO);
Create a FilePlayer audio unit which is a dedicated audio unit to read an audio file and provide audio data in the file to output units, for example, the RemoteIO unit created in step 1. To actually use the FilePlayer, a lot of settings (specify which file to play, which part of the file to play, etc.) are needed to be done on the it;
Set kAudioUnitProperty_SetRenderCallback and kAudioUnitProperty_StreamFormat properties of the RemoteIO unit. The first property is essentially a callback function from which the RemoteIO unit pulls audio data and play it. The second property must be set in accordance to StreamFormat that supported by the FilePlayer. It can be derived from a get-property function invoked on the FilePlayer.
Define the callback set in step 3 where the most important thing to do is asking the FilePlayer to render into the buffer provided by the callback for which you will need to invoke AudioUnitRender() on the FilePlayer.
Finally start the RemoteIO unit to play the file.
Above is just a preliminary outline of basic things to do to play files using audio units on iOS. You can refer to Chris Adamson and Kevin Avila's Learning Core Audio for details.
It's a relatively simple approach that utilizes the audio unit mentioned in the Tasty Pixel blog. In the recording callback, instead of filling the buffer with data from the microphone, you could fill it with data from the file using ExtAudioFileRead. I'll try and paste an example below. Mind you this will just work for .caf files.
In the start method call an readAudio or initAudioFile function, something that just gets all the info about the file.
- (void) start {
readAudio();
OSStatus status = AudioOutputUnitStart(audioUnit);
checkStatus(status);
}
Now in the readAudio method you initialize the audio file reference as such.
ExtAudioFileRef fileRef;
void readAudio() {
NSString * name = #"AudioFile";
NSString * source = [[NSBundle mainBundle] pathForResource:name ofType:#"caf"];
const char * cString = [source cStringUsingEncoding:NSASCIIStringEncoding];
CFStringRef str = CFStringCreateWithCString(NULL, cString, kCFStringEncodingMacRoman);
CFURLRef inputFileURL = CFURLCreateWithFileSystemPath(kCFAllocatorDefault, str, kCFURLPOSIXPathStyle, false);
AudioFileID fileID;
OSStatus err = AudioFileOpenURL(inputFileURL, kAudioFileReadPermission, 0, &fileID);
CheckError(err, "AudioFileOpenURL");
err = ExtAudioFileOpenURL(inputFileURL, &fileRef);
CheckError(err, "ExtAudioFileOpenURL");
err = ExtAudioFileSetProperty(fileRef, kExtAudioFileProperty_ClientDataFormat, sizeof(AudioStreamBasicDescription), &audioFormat);
CheckError(err, "ExtAudioFileSetProperty");
}
Now that you have the Audio Data at hand, next step is pretty easy. In the recordingCallback read the data from the file instead of the mic.
static OSStatus recordingCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
// Because of the way our audio format (setup below) is chosen:
// we only need 1 buffer, since it is mono
// Samples are 16 bits = 2 bytes.
// 1 frame includes only 1 sample
AudioBuffer buffer;
buffer.mNumberChannels = 1;
buffer.mDataByteSize = inNumberFrames * 2;
buffer.mData = malloc( inNumberFrames * 2 );
// Put buffer in a AudioBufferList
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = buffer;
// Then:
// Obtain recorded samples
OSStatus err = ExtAudioFileRead(fileRef, &inNumberFrames, &bufferList);
// Now, we have the samples we just read sitting in buffers in bufferList
// Process the new data
[iosAudio processAudio:&bufferList];
// release the malloc'ed data in the buffer we created earlier
free(bufferList.mBuffers[0].mData);
return noErr;
}
This worked for me.
I'm writing an iOS app that streams video and audio over the network.
I am using AVCaptureSession to grab raw video frames using AVCaptureVideoDataOutput and encode them in software using x264. This works great.
I wanted to do the same for audio, only that I don't need that much control on the audio side so I wanted to use the built in hardware encoder to produce an AAC stream. This meant using Audio Converter from the Audio Toolbox layer. In order to do so I put in a handler for AVCaptudeAudioDataOutput's audio frames:
- (void)captureOutput:(AVCaptureOutput *)captureOutput
didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer
fromConnection:(AVCaptureConnection *)connection
{
// get the audio samples into a common buffer _pcmBuffer
CMBlockBufferRef blockBuffer = CMSampleBufferGetDataBuffer(sampleBuffer);
CMBlockBufferGetDataPointer(blockBuffer, 0, NULL, &_pcmBufferSize, &_pcmBuffer);
// use AudioConverter to
UInt32 ouputPacketsCount = 1;
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0].mNumberChannels = 1;
bufferList.mBuffers[0].mDataByteSize = sizeof(_aacBuffer);
bufferList.mBuffers[0].mData = _aacBuffer;
OSStatus st = AudioConverterFillComplexBuffer(_converter, converter_callback, (__bridge void *) self, &ouputPacketsCount, &bufferList, NULL);
if (0 == st) {
// ... send bufferList.mBuffers[0].mDataByteSize bytes from _aacBuffer...
}
}
In this case the callback function for the audio converter is pretty simple (assuming packet sizes and counts are setup properly):
- (void) putPcmSamplesInBufferList:(AudioBufferList *)bufferList withCount:(UInt32 *)count
{
bufferList->mBuffers[0].mData = _pcmBuffer;
bufferList->mBuffers[0].mDataByteSize = _pcmBufferSize;
}
And the setup for the audio converter looks like this:
{
// ...
AudioStreamBasicDescription pcmASBD = {0};
pcmASBD.mSampleRate = ((AVAudioSession *) [AVAudioSession sharedInstance]).currentHardwareSampleRate;
pcmASBD.mFormatID = kAudioFormatLinearPCM;
pcmASBD.mFormatFlags = kAudioFormatFlagsCanonical;
pcmASBD.mChannelsPerFrame = 1;
pcmASBD.mBytesPerFrame = sizeof(AudioSampleType);
pcmASBD.mFramesPerPacket = 1;
pcmASBD.mBytesPerPacket = pcmASBD.mBytesPerFrame * pcmASBD.mFramesPerPacket;
pcmASBD.mBitsPerChannel = 8 * pcmASBD.mBytesPerFrame;
AudioStreamBasicDescription aacASBD = {0};
aacASBD.mFormatID = kAudioFormatMPEG4AAC;
aacASBD.mSampleRate = pcmASBD.mSampleRate;
aacASBD.mChannelsPerFrame = pcmASBD.mChannelsPerFrame;
size = sizeof(aacASBD);
AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &aacASBD);
AudioConverterNew(&pcmASBD, &aacASBD, &_converter);
// ...
}
This seems pretty straight forward only the IT DOES NOT WORK. Once the AVCaptureSession is running, the audio converter (specifically AudioConverterFillComplexBuffer) returns an 'hwiu' (hardware in use) error. Conversion works fine if the session is stopped but then I can't capture anything...
I was wondering if there was a way to get an AAC stream out of AVCaptureSession. The options I'm considering are:
Somehow using AVAssetWriterInput to encode audio samples into AAC and then get the encoded packets somehow (not through AVAssetWriter, which would only write to a file).
Reorganizing my app so that it uses AVCaptureSession only on the video side and uses Audio Queues on the audio side. This will make flow control (starting and stopping recording, responding to interruptions) more complicated and I'm afraid that it might cause synching problems between the audio and video. Also, it just doesn't seem like a good design.
Does anyone know if getting the AAC out of AVCaptureSession is possible? Do I have to use Audio Queues here? Could this get me into synching or control problems?
I ended up asking Apple for advice (it turns out you can do that if you have a paid developer account).
It seems that AVCaptureSession grabs a hold of the AAC hardware encoder but only lets you use it to write directly to file.
You can use the software encoder but you have to ask for it specifically instead of using AudioConverterNew:
AudioClassDescription *description = [self
getAudioClassDescriptionWithType:kAudioFormatMPEG4AAC
fromManufacturer:kAppleSoftwareAudioCodecManufacturer];
if (!description) {
return false;
}
// see the question as for setting up pcmASBD and arc ASBD
OSStatus st = AudioConverterNewSpecific(&pcmASBD, &aacASBD, 1, description, &_converter);
if (st) {
NSLog(#"error creating audio converter: %s", OSSTATUS(st));
return false;
}
with
- (AudioClassDescription *)getAudioClassDescriptionWithType:(UInt32)type
fromManufacturer:(UInt32)manufacturer
{
static AudioClassDescription desc;
UInt32 encoderSpecifier = type;
OSStatus st;
UInt32 size;
st = AudioFormatGetPropertyInfo(kAudioFormatProperty_Encoders,
sizeof(encoderSpecifier),
&encoderSpecifier,
&size);
if (st) {
NSLog(#"error getting audio format propery info: %s", OSSTATUS(st));
return nil;
}
unsigned int count = size / sizeof(AudioClassDescription);
AudioClassDescription descriptions[count];
st = AudioFormatGetProperty(kAudioFormatProperty_Encoders,
sizeof(encoderSpecifier),
&encoderSpecifier,
&size,
descriptions);
if (st) {
NSLog(#"error getting audio format propery: %s", OSSTATUS(st));
return nil;
}
for (unsigned int i = 0; i < count; i++) {
if ((type == descriptions[i].mSubType) &&
(manufacturer == descriptions[i].mManufacturer)) {
memcpy(&desc, &(descriptions[i]), sizeof(desc));
return &desc;
}
}
return nil;
}
The software encoder will take up CPU resources, of course, but will get the job done.