【问题标题】:Duplex Audio communication using AudioUnits使用 AudioUnits 进行双工音频通信
【发布时间】:2014-03-21 14:58:30
【问题描述】:

我正在开发一个具有以下要求的应用程序:

  1. 从 iOS 设备 (iPhone/iPad) 录制实时音频并通过网络发送到服务器
  2. 在 iOS 设备(iPhone/iPad)上播放从网络服务器接收到的音频

上面提到的事情需要同时完成。

我为此使用了AudioUnit

我遇到了一个问题,我听到的音频与我对 iPhone Mic 说话的音频相同,而不是从网络服务器接收到的音频。

我已经搜索了很多关于如何避免这种情况但没有找到解决方案的方法。

如果有人遇到同样的问题并找到任何解决方案,分享会很有帮助。

这是我初始化音频单元的代码

-(void)initializeAudioUnit
{

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

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

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


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

    status = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO,
                                kAudioUnitScope_Output,
                                kOutputBus,
                                &flag,
                                sizeof(flag));


    AudioStreamBasicDescription audioStreamBasicDescription;

    // Describe format
    audioStreamBasicDescription.mSampleRate         = 16000;
    audioStreamBasicDescription.mFormatID           = kAudioFormatLinearPCM;
    audioStreamBasicDescription.mFormatFlags        = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked |kLinearPCMFormatFlagIsNonInterleaved;
    audioStreamBasicDescription.mFramesPerPacket    = 1;
    audioStreamBasicDescription.mChannelsPerFrame   = 1;
    audioStreamBasicDescription.mBitsPerChannel     = 16;
    audioStreamBasicDescription.mBytesPerPacket     = 2;
    audioStreamBasicDescription.mBytesPerFrame      = 2;



    status = AudioUnitSetProperty(audioUnit,
                              kAudioUnitProperty_StreamFormat,
                              kAudioUnitScope_Output,
                              kInputBus,
                              &audioStreamBasicDescription,
                              sizeof(audioStreamBasicDescription));
    NSLog(@"Status[%d]",(int)status);


status = AudioUnitSetProperty(audioUnit,
                                kAudioUnitProperty_StreamFormat,
                                kAudioUnitScope_Input,
                                kOutputBus,
                                &audioStreamBasicDescription,
                                sizeof(audioStreamBasicDescription));
NSLog(@"Status[%d]",(int)status);


    AURenderCallbackStruct callbackStruct;


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

  callbackStruct.inputProc = playbackCallback;
      callbackStruct.inputProcRefCon = (__bridge void *)(self);
  status = AudioUnitSetProperty(audioUnit,
                                kAudioUnitProperty_SetRenderCallback,
                                kAudioUnitScope_Global,
                                kOutputBus,
                                &callbackStruct,
                                sizeof(callbackStruct));
    flag=0;

status = AudioUnitSetProperty(audioUnit,
                                kAudioUnitProperty_ShouldAllocateBuffer,
                                kAudioUnitScope_Output,
                                kInputBus,
                                &flag,
                                sizeof(flag));

}

录音回拨

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

    AudioBuffer tempBuffer;
    tempBuffer.mNumberChannels = 1;
    tempBuffer.mDataByteSize = inNumberFrames * 2;
    tempBuffer.mData = malloc(inNumberFrames *2);

    AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = tempBuffer;





    OSStatus status;
    status = AudioUnitRender(THIS->audioUnit,
                         ioActionFlags,
                         inTimeStamp,
                         kInputBus,
                         inNumberFrames,
                         &bufferList);

    if (noErr != status) {

        printf("AudioUnitRender error: %d", (int)status);
        return noErr;
    }

    tempBuffer.mDataByteSize, &encodedSize,(__bridge void *)(THIS));

    [THIS processAudio:&bufferList];

    free(bufferList.mBuffers[0].mData);

    return noErr;
}

回放回调

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


NSLog(@"In play back call back");


MyAudioViewController *THIS = (__bridge MyAudioViewController *)inRefCon;



int32_t availableBytes=0;


  char *inBuffer = GetDataFromCircularBuffer(&THIS->mybuffer, &availableBytes);
  NSLog(@"bytes available in buffer[%d]",availableBytes);
  decodeSpeexData(inBuffer, availableBytes,(__bridge void *)(THIS));
  ConsumeReadBytes(&(THIS->mybuffer), availableBytes); 

  memcpy(targetBuffer, THIS->outTemp, inNumberFrames*2);


 return noErr;
}

处理从 MIC 录制的音频

- (void) processAudio: (AudioBufferList*) bufferList
{
    AudioBuffer sourceBuffer = bufferList->mBuffers[0];

    //    NSLog(@"Origin size: %d", (int)sourceBuffer.mDataByteSize);
    int size = 0;
    encodeAudioDataSpeex((spx_int16_t*)sourceBuffer.mData, sourceBuffer.mDataByteSize, &size, (__bridge void *)(self));
    [self performSelectorOnMainThread:@selector(SendAudioData:) withObject:[NSData dataWithBytes:self->jitterBuffer length:size] waitUntilDone:NO];

    NSLog(@"Encoded size: %i", size);

} 

【问题讨论】:

    标签: ios core-audio record playback audiounit


    【解决方案1】:

    您未显示的播放回调渲染回调负责发送到 RemoteIO 扬声器输出的音频。如果此 RemoteIO 渲染回调在其回调缓冲区中没有放置任何数据,则缓冲区中留下的任何垃圾(可能之前在记录回调缓冲区中的东西)可能会被发送到扬声器。

    此外,Apple DTS 强烈建议您的 recordingCallback 不包括 任何 内存管理调用,例如 malloc()。因此,这也可能是导致问题的错误。

    【讨论】:

    • 感谢您指出可能的错误。我会纠正它。我正在发布我的播放回调。我想知道为什么我听到我正在捕捉的东西。我只想播放从网络服务器接收到的音频。
    • 我注意到,如果我在录音回叫中注释掉 ProcessAudio 呼叫,我不会听到我对着麦克风讲话的任何内容。可能是什么原因?
    • 您的录制音频回调可能需要很长时间。不应在实时音频单元回调中进行扩展处理、锁定或内存管理。只有时间有界的东西比样本数除以采样率要快得多,例如将 memcpy 放入循环 FIFO 或队列。
    • 感谢您的指出。我已经完成了更改,现在它可以工作了。
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