Сжатие звука в реальном времени с помощью AudioConverterFillComplexBuffer в iOS

Я пытаюсь выполнить кодирование в реальном времени буфера, поступающего от записи обратного вызова, но я, кажется, не совсем понимаю, как это сделать и как это работает. На самом деле я часами читал ссылку Apple, но я просто не понимаю.

Я хочу просмотреть различные темы здесь, на SO, но все равно мне это не подходит.

У меня есть обратный вызов записи:

static OSStatus recordingCallback(void *inRefCon,
                              AudioUnitRenderActionFlags *ioActionFlags,
                              const AudioTimeStamp *inTimeStamp,
                              UInt32 inBusNumber,
                              UInt32 inNumberFrames,
                              AudioBufferList *ioData) {

// the data gets rendered here
AudioBuffer buffer;

// a variable where we check the status
OSStatus status;

/**
 This is the reference to the object who owns the callback.
 */
AudioProcessor *audioProcessor = (__bridge AudioProcessor*) inRefCon;

/**
 on this point we define the number of channels, which is mono
 for the iphone. the number of frames is usally 512 or 1024.
 */
buffer.mDataByteSize = inNumberFrames * 2; // sample size
buffer.mNumberChannels = 1; // one channel
buffer.mData = malloc( inNumberFrames * 2 ); // buffer size

// we put our buffer into a bufferlist array for rendering
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = buffer;

// render input and check for error
status = AudioUnitRender([audioProcessor audioUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList);
[audioProcessor hasError:status:__FILE__:__LINE__];

// process the bufferlist in the audio processor
[audioProcessor processBuffer:&bufferList];

[audioProcessor convertBuffer:&bufferList]; //trying to set the buffer to converting function

// clean up the buffer
free(bufferList.mBuffers[0].mData);

return noErr;
}

convertBuffer выглядит так — он возвращает

освобождаемый указатель не был выделен

-(void)convertBuffer: (AudioBufferList*) audioBufferList
{
    unsigned char** stream = audioBufferList->mBuffers[0].mData;
    UInt32 *streamSize = &audioBufferList->mBuffers[0].mDataByteSize;
    // describe the input format's description

    AudioStreamBasicDescription inputDescription;
    inputDescription.mSampleRate            = SAMPLE_RATE;
    inputDescription.mFormatID          = kAudioFormatLinearPCM;
    inputDescription.mFormatFlags       = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;
    inputDescription.mFramesPerPacket   = 1;
    inputDescription.mChannelsPerFrame  = 1;
    inputDescription.mBitsPerChannel        = 16;
    inputDescription.mBytesPerPacket        = 2;
    inputDescription.mBytesPerFrame     = 2;

    // copy conversion output format's description from the
    // output audio unit's description.
    // then adjust framesPerPacket to match the input we'll be passing.

    // framecount of our input stream is based on the input bytecount.
    // output stream will have same number of frames, but different
    // number of bytes.
    AudioStreamBasicDescription outputDescription;
    outputDescription.mSampleRate         = 44100.0;
    outputDescription.mFormatID           = kAudioFormatMPEG4AAC;
    outputDescription.mFormatFlags        = kMPEG4Object_AAC_Main;
    outputDescription.mChannelsPerFrame   = 1;
    outputDescription.mBytesPerPacket     = 1;
    outputDescription.mBytesPerFrame      = 0;
    outputDescription.mFramesPerPacket    = 1024;
    outputDescription.mBitsPerChannel     = 0;
    outputDescription.mReserved           = 0;


    // create an audio converter
    AudioConverterRef audioConverter;
    OSStatus acCreationResult = AudioConverterNew(&inputDescription, &outputDescription, &audioConverter);
    if(!audioConverter)
    {
        // bail out
        free(*stream);
        *streamSize = 0;
        *stream = (unsigned char*)malloc(0);
        return;
    }

    // calculate number of bytes required for output of input stream.
    // allocate buffer of adequate size.
    UInt32 outputBytes = outputDescription.mBytesPerPacket * (*streamSize / inputDescription.mBytesPerPacket); // outputDescription.mFramesPerPacket * outputDescription.mBytesPerFrame;
    unsigned char *outputBuffer = (unsigned char*)malloc(outputBytes);
    memset(outputBuffer, 0, outputBytes);

    // describe input data we'll pass into converter
    AudioBuffer inputBuffer;
    inputBuffer.mNumberChannels = inputDescription.mChannelsPerFrame;
    inputBuffer.mDataByteSize = *streamSize;
    inputBuffer.mData = *stream;

    // describe output data buffers into which we can receive data.
    AudioBufferList outputBufferList;
    outputBufferList.mNumberBuffers = 1;
    outputBufferList.mBuffers[0].mNumberChannels = outputDescription.mChannelsPerFrame;
    outputBufferList.mBuffers[0].mDataByteSize = outputBytes;
    outputBufferList.mBuffers[0].mData = outputBuffer;

    // set output data packet size
    UInt32 outputDataPacketSize = outputBytes / outputDescription.mBytesPerPacket;

    // fill class members with data that we'll pass into
    // the InputDataProc
    //_converter_currentBuffer = &inputBuffer;
    //_converter_currentInputDescription = inputDescription;

    // convert
    OSStatus result = AudioConverterFillComplexBuffer(audioConverter, /* AudioConverterRef inAudioConverter */
                                                      nil, /* AudioConverterComplexInputDataProc inInputDataProc */
                                                      nil, /* void *inInputDataProcUserData */
                                                      &outputDataPacketSize, /* UInt32 *ioOutputDataPacketSize */
                                                      &outputBufferList, /* AudioBufferList *outOutputData */
                                                      NULL /* AudioStreamPacketDescription *outPacketDescription */
                                                      );

NSLog([NSString stringWithFormat:@"Err : %ld",result ]);

    // change "stream" to describe our output buffer.
    // even if error occured, we'd rather have silence than unconverted audio.
    free(*stream);
    *stream = outputBuffer;
    *streamSize = outputBytes;

    // dispose of the audio converter
    AudioConverterDispose(audioConverter);
}

и в основном это все. Не могу заставить его работать. Кроме того, как должна выглядеть функция обратного вызова? Я установил его на ноль на данный момент.

Любая помощь высоко ценится. Я застрял в нем на несколько дней.