ios - 出力なしのリアルタイムオーディオ処理

Question

私はこの例を見ていますhttp://teragonaudio.com/article/How-to-do-realtime-recording-with-effect-processing-on-iOS.html

出力をオフにしたい。に変更しようとしkAudioSessionCategory_PlayAndRecordましたkAudioSessionCategory_RecordAudioが、これは機能していません。私も取り除こうとします：

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

マイクから音を出したいのですが、再生していません。しかし、サウンドが renderCallback メソッドに到達したときに何をしても、-50 エラーが発生します。オーディオが出力で自動的に再生されると、すべて正常に動作します...

コードで更新：

using namespace std;

AudioUnit *audioUnit = NULL;

float *convertedSampleBuffer = NULL;

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

    if(AudioSessionInitialize(NULL, 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,
                               sizeof(UInt32), &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,
                                      1, 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,
                               sizeof(UInt32), &audioCategory) != noErr) {
        return 1;
    }

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

    AURenderCallbackStruct callbackStruct;
    callbackStruct.inputProc = renderCallback; // Render function
    callbackStruct.inputProcRefCon = NULL;
    if(AudioUnitSetProperty(*audioUnit, kAudioUnitProperty_SetRenderCallback,
                            kAudioUnitScope_Input, 0, &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, kAudioUnitProperty_StreamFormat,
                            kAudioUnitScope_Input, 0, &streamDescription, sizeof(streamDescription)) != noErr) {
        return 1;
    }

    // Ditto for the output stream, which we will be sending the processed audio to
    if(AudioUnitSetProperty(*audioUnit, kAudioUnitProperty_StreamFormat,
                            kAudioUnitScope_Output, 1, &streamDescription, 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;
}

そして私の VC からの呼び出し:

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

Answer 1

記録のみが必要で、再生は必要ない場合は、renderCallback を設定する行をコメント アウトするだけです。

AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = renderCallback; // Render function
callbackStruct.inputProcRefCon = NULL;
if(AudioUnitSetProperty(*audioUnit, kAudioUnitProperty_SetRenderCallback,
   kAudioUnitScope_Input, 0, &callbackStruct,
   sizeof(AURenderCallbackStruct)) != noErr) {
  return 1;
}

コードを見た後の更新：

私が疑ったように、入力コールバックがありません。次の行を追加します。

// at top:
#define kInputBus 1

AURenderCallbackStruct callbackStruct;
/**/
callbackStruct.inputProc = &ALAudioUnit::recordingCallback;
callbackStruct.inputProcRefCon = this;
status = AudioUnitSetProperty(audioUnit,
                              kAudioOutputUnitProperty_SetInputCallback,
                              kAudioUnitScope_Global,
                              kInputBus,
                              &callbackStruct,
                              sizeof(callbackStruct));

今あなたのrecordingCallbackで：

OSStatus ALAudioUnit::recordingCallback(void *inRefCon,
                                        AudioUnitRenderActionFlags *ioActionFlags,
                                        const AudioTimeStamp *inTimeStamp,
                                        UInt32 inBusNumber,
                                        UInt32 inNumberFrames,
                                        AudioBufferList *ioData)
{
    // TODO: Use inRefCon to access our interface object to do stuff
    // Then, use inNumberFrames to figure out how much data is available, and make
    // that much space available in buffers in an AudioBufferList.

    // Then:
    // Obtain recorded samples

    OSStatus status;

    ALAudioUnit *pThis = reinterpret_cast<ALAudioUnit*>(inRefCon);
    if (!pThis)
        return noErr;

    //assert (pThis->m_nMaxSliceFrames >= inNumberFrames);

    pThis->recorderBufferList->GetBufferList().mBuffers[0].mDataByteSize = inNumberFrames * pThis->m_recorderSBD.mBytesPerFrame;

    status = AudioUnitRender(pThis->audioUnit,
                             ioActionFlags,
                             inTimeStamp,
                             inBusNumber,
                             inNumberFrames,
                             &pThis->recorderBufferList->GetBufferList());
    THROW_EXCEPTION_IF_ERROR(status, "error rendering audio unit");

    // If we're not playing, I don't care about the data, simply discard it
    if (!pThis->playbackState || pThis->isSeeking) return noErr;

    // Now, we have the samples we just read sitting in buffers in bufferList
    pThis->DoStuffWithTheRecordedAudio(inNumberFrames, pThis->recorderBufferList, inTimeStamp);

    return noErr;
}

ところで、AudioUnit が提供するものを使用する代わりに、独自のバッファーを割り当てています。AudioUnit に割り当てられたバッファーを使用する場合は、これらの部分を変更することをお勧めします。

アップデート：

独自のバッファを割り当てる方法:

recorderBufferList = new AUBufferList();
recorderBufferList->Allocate(m_recorderSBD, m_nMaxSliceFrames);
recorderBufferList->PrepareBuffer(m_recorderSBD, m_nMaxSliceFrames);

また、これを行う場合は、AudioUnit にバッファーを割り当てないように指示します。

// 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_Input,
                              kInputBus,
                              &flag,
                              sizeof(flag));

CoreAudio ユーティリティ クラスを含める必要があります。

Answer 2

@ Mar0ux の回答に感謝します。これを行う完全なサンプルコードを探している人は誰でも、ここで見ることができます:

https://code.google.com/p/ios-coreaudio-example/

Answer 3

私は同じコードで同様のアプリを実行していますが、列挙kAudioSessionCategory_PlayAndRecordを次のように変更することで再生を終了できることがわかりました。RecordAudio

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

これにより、ハードウェアのマイクとスピーカーの間のフィードバックが停止しました。