我想关掉我的输出.我尝试将kAudioSessionCategory_PlayAndRecord更改为kAudioSessionCategory_RecordAudio,但这不起作用.我也试图摆脱：

if(AudioUnitSetProperty(*audioUnit,kAudioUnitProperty_StreamFormat,kAudioUnitScope_Output,1,&streamDescription,sizeof(streamDescription)) != noErr) {
        return 1;
    }

Becouse我想从麦克风那里获得声音但不能播放它.但是当我的声音变为renderCallback方法时,无论我做什么都有-50错误.当音频在输出上自动播放时,一切正常……

使用代码更新：

using namespace std;

AudioUnit *audioUnit = NULL;

float *convertedSampleBuffer = NULL;

int initAudioSession() {
    audioUnit = (AudioUnit*)malloc(sizeof(AudioUnit));

    if(AudioSessionInitialize(NULL,NULL,NULL) != noErr) {
        return 1;
    }

    if(AudioSessionSetActive(true) != noErr) {
        return 1;
    }

    UInt32 sessionCategory = kAudioSessionCategory_PlayAndRecord;
    if(AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,sizeof(UInt32),&sessionCategory) != noErr) {
        return 1;
    }

    Float32 bufferSizeInSec = 0.02f;
    if(AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration,sizeof(Float32),&bufferSizeInSec) != noErr) {
        return 1;
    }

    UInt32 overrideCategory = 1;
    if(AudioSessionSetProperty(kAudioSessionProperty_OverrideCategoryDefaultToSpeaker,&overrideCategory) != noErr) {
        return 1;
    }

    // There are many properties you might want to provide callback functions for:
    // kAudioSessionProperty_AudioRouteChange
    // kAudioSessionProperty_OverrideCategoryEnableBluetoothInput
    // etc.

    return 0;
}

OSStatus renderCallback(void *userData,AudioUnitRenderActionFlags *actionFlags,const AudioTimeStamp *audioTimeStamp,UInt32 busNumber,UInt32 numFrames,AudioBufferList *buffers) {
    OSStatus status = AudioUnitRender(*audioUnit,actionFlags,audioTimeStamp,numFrames,buffers);

    int doOutput = 0;

    if(status != noErr) {
        return status;
    }

    if(convertedSampleBuffer == NULL) {
        // Lazy initialization of this buffer is necessary because we don't
        // know the frame count until the first callback
        convertedSampleBuffer = (float*)malloc(sizeof(float) * numFrames);
        baseTime = (float)QRealTimer::getUptimeInMilliseconds();
    }

    SInt16 *inputFrames = (SInt16*)(buffers->mBuffers->mData);

    // If your DSP code can use integers,then don't bother converting to
    // floats here,as it just wastes cpu. However,most DSP algorithms rely
    // on floating point,and this is especially true if you are porting a
    // VST/AU to iOS.

    int i;

    for( i = numFrames; i < fftlength; i++ )        // Shifting buffer
        x_inbuf[i - numFrames] = x_inbuf[i];

    for(  i = 0; i < numFrames; i++) {
        x_inbuf[i + x_phase] = (float)inputFrames[i] / (float)32768;
    }

    if( x_phase + numFrames == fftlength )
    {
        x_alignment.SigProc_frontend(x_inbuf);  // Signal processing front-end (FFT!)
        doOutput = x_alignment.Align();


        /// Output as text! In the real-time version,//      this is where we update visualisation callbacks and launch other services
        if ((doOutput) & (x_netscore.isEvent(x_alignment.Position()))
            &(x_alignment.lastAction()<x_alignment.Position()) )
        {
          //  here i want to do something with my input!
        }
    }
    else
        x_phase += numFrames;


   return noErr;
}


int initAudioStreams(AudioUnit *audioUnit) {
    UInt32 audioCategory = kAudioSessionCategory_PlayAndRecord;
    if(AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,&audioCategory) != noErr) {
        return 1;
    }

    UInt32 overrideCategory = 1;
    if(AudioSessionSetProperty(kAudioSessionProperty_OverrideCategoryDefaultToSpeaker,&overrideCategory) != noErr) {
        // Less serIoUs error,but you may want to handle it and bail here
    }

    AudioComponentDescription componentDescription;
    componentDescription.componentType = kAudioUnitType_Output;
    componentDescription.componentSubType = kAudioUnitSubType_RemoteIO;
    componentDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
    componentDescription.componentFlags = 0;
    componentDescription.componentFlagsMask = 0;
    AudioComponent component = AudioComponentFindNext(NULL,&componentDescription);
    if(AudioComponentInstanceNew(component,audioUnit) != noErr) {
        return 1;
    }

    UInt32 enable = 1;
    if(AudioUnitSetProperty(*audioUnit,kAudioOutputUnitProperty_EnableIO,kAudioUnitScope_Input,&enable,sizeof(UInt32)) != noErr) {
        return 1;
    }

    AURenderCallbackStruct callbackStruct;
    callbackStruct.inputProc = renderCallback; // Render function
    callbackStruct.inputProcRefCon = NULL;
    if(AudioUnitSetProperty(*audioUnit,kAudioUnitProperty_SetRenderCallback,&callbackStruct,sizeof(AURenderCallbackStruct)) != noErr) {
        return 1;
    }

    AudioStreamBasicDescription streamDescription;
    // You might want to replace this with a different value,but keep in mind that the
    // iPhone does not support all sample rates. 8kHz,22kHz,and 44.1kHz should all work.
    streamDescription.mSampleRate = 44100;
    // Yes,I know you probably want floating point samples,but the iPhone isn't going
    // to give you floating point data. You'll need to make the conversion by hand from
    // linear PCM <-> float.
    streamDescription.mFormatID = kAudioFormatLinearPCM;
    // This part is important!
    streamDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger |
    kAudioFormatFlagsNativeEndian |
    kAudioFormatFlagIsPacked;
    streamDescription.mBitsPerChannel = 16;
    // 1 sample per frame,will always be 2 as long as 16-bit samples are being used
    streamDescription.mBytesPerFrame = 2;
    streamDescription.mChannelsPerFrame = 1;
    streamDescription.mBytesPerPacket = streamDescription.mBytesPerFrame *
    streamDescription.mChannelsPerFrame;
    // Always should be set to 1
    streamDescription.mFramesPerPacket = 1;
    // Always set to 0,just to be sure
    streamDescription.mReserved = 0;

    // Set up input stream with above properties
    if(AudioUnitSetProperty(*audioUnit,sizeof(streamDescription)) != noErr) {
        return 1;
    }

    // Ditto for the output stream,which we will be sending the processed audio to
    if(AudioUnitSetProperty(*audioUnit,sizeof(streamDescription)) != noErr) {
        return 1;
    }

    return 0;
}


int startAudioUnit(AudioUnit *audioUnit) {
    if(AudioUnitInitialize(*audioUnit) != noErr) {
        return 1;
    }

    if(AudioOutputUnitStart(*audioUnit) != noErr) {
        return 1;
    }

    return 0;
}

从我的VC打电话：

initAudioSession();
    initAudioStreams( audioUnit);
    startAudioUnit( audioUnit);