AVAudioSession输出为A2DP时强制输入输入

Question

我想将输入从line-in重新路由到A2DP。但遗憾的是，我无法将输入设置为输入。有没有办法强制输入输入然后输出到A2DP？

下面的配置。请注意“set line in as preferred”附近的代码。我相信这应该正确设置输入 - 如果它可用...

- (id) init {
    self = [super init];

    sizeof(allowBluetoothInput), &allowBluetoothInput);


    // You can adjust the latency of RemoteIO (and, in fact, any other audio framework) by setting the kAudioSessionProperty_PreferredHardwareIOBufferDuration property
    float aBufferLength = 0.005; // In seconds
    AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof(aBufferLength), &aBufferLength);

    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;


    // set line in as preferred
    AVAudioSession *audioSession = [AVAudioSession sharedInstance];
    NSString *preferredPortType = AVAudioSessionPortLineIn;
    for (AVAudioSessionPortDescription *desc in      audioSession.availableInputs) {
         if ([desc.portType isEqualToString: AVAudioSessionPortLineIn] ||
        //            [desc.portType isEqualToString: AVAudioSessionPortBuiltInMic] ||
        [desc.portType isEqualToString: AVAudioSessionPortHeadsetMic])
        {
            [audioSession setPreferredInput:desc error:nil];

        }
    }

    // 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         = 44100.00;
    audioFormat.mFormatID           = kAudioFormatLinearPCM; // kAudioFormatMPEG4AAC_ELD
    audioFormat.mFormatFlags        = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
    audioFormat.mFramesPerPacket    = 1;
    audioFormat.mChannelsPerFrame   = 1;
    audioFormat.mBitsPerChannel     = 16;
    audioFormat.mBytesPerPacket     = 2;
    audioFormat.mBytesPerFrame      = 2;

    // 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 = self;
    status = AudioUnitSetProperty(audioUnit, 
                                  kAudioOutputUnitProperty_SetInputCallback, 
                                  kAudioUnitScope_Global, 
                                  kInputBus, 
                                  &callbackStruct, 
                                  sizeof(callbackStruct));
    checkStatus(status);

    // Set output callback
    callbackStruct.inputProc = playbackCallback;
    callbackStruct.inputProcRefCon = 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 );

    [[AVAudioSession sharedInstance] setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:AVAudioSessionCategoryOptionAllowBluetoothA2DP error:NULL];

    // Initialise
    AudioSessionSetActive(true);

    NSTimeInterval outLat = [[AVAudioSession sharedInstance] outputLatency];
    NSTimeInterval inLat = [[AVAudioSession sharedInstance] inputLatency];

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

    return self;
}