Question

I just try to developing a VOIP application,

• the audio buffer which fetch from RecordingCallBack would be wrapped
  to a NSData and then send to the remote-side by GCDAsyncSocket

• and the remote-side would get the NSData, unwrapped to an audio
  buffer, and then the PlayingCallBack will fetch the audio buffer.

my plan is working so far, running fine on local ( the socket send data to local, and play the buffer local )

but when it running on two devices ( one real iphone-4s, one simulator )
the voice would became stranger, sounds like robotic sound

is there anyway to avoid the robotic sound effect ?

Here is my AudioUnit Settings:

#pragma mark - Init Methods

- (void)initAudioUint
{
    OSStatus status;

    // Describe audio component
    AudioComponentDescription desc;
    desc.componentType = kAudioUnitType_Output;
    desc.componentSubType = kAudioUnitSubType_RemoteIO;
    desc.componentFlags = 0;
    desc.componentFlagsMask = 0;
    desc.componentManufacturer = kAudioUnitManufacturer_Apple;

    // Get component
    AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);

    // Get audio units
    status = AudioComponentInstanceNew(inputComponent, &audioUnit);
    checkStatus(status);

    // Enable IO for recording
    UInt32 flag = 1;
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioOutputUnitProperty_EnableIO,
                                  kAudioUnitScope_Input,
                                  kInputBus,
                                  &flag,
                                  sizeof(flag));
    checkStatus(status);

    // Enable IO for playback
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioOutputUnitProperty_EnableIO,
                                  kAudioUnitScope_Output,
                                  kOutputBus,
                                  &flag,
                                  sizeof(flag));
    checkStatus(status);

    // Describe format
    AudioStreamBasicDescription audioFormat;
    audioFormat.mSampleRate = 44100.0f; // FS
    audioFormat.mFormatID = kAudioFormatLinearPCM;
    audioFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
    audioFormat.mChannelsPerFrame = 1; // stereo output
    audioFormat.mFramesPerPacket = 1;
    audioFormat.mBitsPerChannel = sizeof(short) * 8; // 16-bit
    audioFormat.mBytesPerFrame = audioFormat.mBitsPerChannel / 8 * audioFormat.mChannelsPerFrame;
    audioFormat.mBytesPerPacket = audioFormat.mBytesPerFrame * audioFormat.mFramesPerPacket;

    // Apply format
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioUnitProperty_StreamFormat,
                                  kAudioUnitScope_Output,
                                  kInputBus,
                                  &audioFormat,
                                  sizeof(audioFormat));
    checkStatus(status);
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioUnitProperty_StreamFormat,
                                  kAudioUnitScope_Input,
                                  kOutputBus,
                                  &audioFormat,
                                  sizeof(audioFormat));
    checkStatus(status);


    // Set input callback
    AURenderCallbackStruct callbackStruct;
    callbackStruct.inputProc = recordingCallback;
    callbackStruct.inputProcRefCon = (__bridge void*)self;
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioOutputUnitProperty_SetInputCallback,
                                  kAudioUnitScope_Global,
                                  kInputBus,
                                  &callbackStruct,
                                  sizeof(callbackStruct));
    checkStatus(status);


    // Set output callback
    callbackStruct.inputProc = playbackCallback;
    callbackStruct.inputProcRefCon = (__bridge void*)self;
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioUnitProperty_SetRenderCallback,
                                  kAudioUnitScope_Global,
                                  kOutputBus,
                                  &callbackStruct,
                                  sizeof(callbackStruct));
    checkStatus(status);


    /*
    // Disable buffer allocation for the recorder (optional - do this if we want to pass in our own)
    flag = 0;
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioUnitProperty_ShouldAllocateBuffer,
                                  kAudioUnitScope_Output,
                                  kInputBus,
                                  &flag,
                                  sizeof(flag));

    // Allocate our own buffers (1 channel, 16 bits per sample, thus 16 bits per frame, thus 2 bytes per frame).
    // Practice learns the buffers used contain 512 frames, if this changes it will be fixed in processAudio.
    tempBuffer.mNumberChannels = 1;
    tempBuffer.mDataByteSize = 512 * 2;
    tempBuffer.mData = malloc( 512 * 2 );
    checkStatus(status);


    // Disable buffer allocation for the recorder (optional - do this if we want to pass in our own)
    flag = 0;
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioUnitProperty_ShouldAllocateBuffer,
                                  kAudioUnitScope_Output,
                                  kInputBus,
                                  &flag,
                                  sizeof(flag));

    // TODO: Allocate our own buffers if we want
    */

    // Initialise
    status = AudioUnitInitialize(audioUnit);
    checkStatus(status);

    conversionBuffer = (SInt16 *) malloc(1024 * sizeof(SInt16));
}

BTW, is there any way to set the audioFormat.mFramesPerPacket > 1 ?

in my case, it would print error, if the param > 1.

I was thinking about send a buffer which contain multi-frames (for
fetch more time to play on the remote-side), it should be better than
send one frame one packet for VOIP ?

Answer 1

I just resolved this problem now!

need to set up the property of audio session, make sure two device has same BufferDuration

    // set preferred buffer size
    Float32 audioBufferSize = (set up the duration);
    UInt32 size = sizeof(audioBufferSize);
    result = AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration,
                           size, &audioBufferSize);