iOS:如何在运行时使用audio Unit重新采样音频(PCM数据)



如何在运行时/现场使用音频单元重新采样音频(PCM数据)?

我有一个音频单元设置如下。

- (void) setUpAudioUnit {
    OSStatus status;
    AudioComponentInstance audioUnit;
    AudioComponent inputComponent;
    AudioComponentDescription audioComponentDescription;
    AudioStreamBasicDescription audioStreamBasicDescription;
    // Describe audio component
    audioComponentDescription.componentType = kAudioUnitType_Output;
    audioComponentDescription.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
    audioComponentDescription.componentFlags = 0;
    audioComponentDescription.componentFlagsMask = 0;
    audioComponentDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
    // Get component
    inputComponent = AudioComponentFindNext(NULL, &audioComponentDescription);
    // 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.mSampleRate         = AUDIO_SAMPLE_RATE;
    audioStreamBasicDescription.mFormatID           = kAudioFormatLinearPCM;
    audioStreamBasicDescription.mFormatFlags        = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
    audioStreamBasicDescription.mFramesPerPacket    = AUDIO_FRAMES_PER_PACKET;
    audioStreamBasicDescription.mChannelsPerFrame   = AUDIO_CHANNELS_PER_FRAME;
    audioStreamBasicDescription.mBitsPerChannel     = AUDIO_BITS_PER_CHANNEL;
    audioStreamBasicDescription.mBytesPerPacket     = AUDIO_BYTES_PER_PACKET;
    audioStreamBasicDescription.mBytesPerFrame      = AUDIO_BYTES_PER_FRAME;
    // Apply format
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioUnitProperty_StreamFormat,
                                  kAudioUnitScope_Output,
                                  kInputBus,
                                  &audioStreamBasicDescription,
                                  sizeof(audioStreamBasicDescription));
    checkStatus(status);

    /* Make sure we set the correct audio category before restarting */
    UInt32 audioCategory = kAudioSessionCategory_PlayAndRecord;
    status = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,
                                     sizeof(audioCategory),
                                     &audioCategory);
    checkStatus(status);

    status = AudioUnitSetProperty(audioUnit,
                                  kAudioUnitProperty_StreamFormat,
                                  kAudioUnitScope_Input,
                                  kOutputBus,
                                  &audioStreamBasicDescription,
                                  sizeof(audioStreamBasicDescription));
    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));

}

音频设置如下:

kOutputBus 0
kInputBus 1
AUDIO_SAMPLE_RATE 44100
AUDIO_FRAMES_PER_PACKET 1
AUDIO_CHANNELS_PER_FRAME 1
AUDIO_BITS_PER_CHANNEL 16 
AUDIO_BYTES_PER_PACKET 2
AUDIO_BYTES_PER_FRAME 2

我正在接收来自记录回调的PCM数据

audioBufferList->mBuffers[0].mData

那么,我如何从44.1KHz到8KHz重新采样这个PCM数据,反之亦然?我已经谷歌了很多,但没有找到任何代码样本或直接的指令。

找到了这些线程,但是没有一个提供明确的指令。

  1. 哪个内置的AudioUnit可以重新采样音频?
  2. 在iOS上改变AUGraph的采样率

任何代码示例或信息,高度赞赏

转换器音频单元将处理您的采样率转换。我发现解决这个问题的最佳方法是调整您的链以适应硬件的本地功能。这意味着您应该获得系统AudioStreamBasicDescription (sysASBD),然后在系统和链中需要不同内容的部分之间放置转换器单元。因此,对于音频播放通过8K采样,你会这样做:ReomoteIO(mic) ->转换器-> your8Kprocessing ->转换器-> RemoteIO(out)。

这是转换器的描述。

AudioComponentDescription convDesc;
convDesc.componentType = kAudioUnitType_FormatConverter;
convDesc.componentSubType = kAudioUnitSubType_AUConverter;
convDesc.componentFlags = 0;
convDesc.componentFlagsMask = 0;
convDesc.componentManufacturer = kAudioUnitManufacturer_Apple;

这是如何获得系统ASBDin和ASBDout

UInt32 sizeASBD = sizeof(AudioStreamBasicDescription);
AudioStreamBasicDescription ioASBDin;
AudioStreamBasicDescription ioASBDout;
AudioUnitGetProperty(remoteIO, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &ioASBDin, &sizeASBD);
AudioUnitGetProperty(remoteIO, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &ioASBDout, &sizeASBD);

使用转换器所要做的就是将它的输入ASBD和输出ASBD设置为所需的格式,它完成所有的工作。连接好你的网络,你就可以玩8K了。

AudioStreamBasicDescription asbd8K;
AudioComponentInstance converter44To8;
AudioUnitSetProperty(converter44To8,kAudioUnitProperty_StreamFormat,kAudioUnitScope_Input,0,& ioASBDin,sizeof(AudioStreamBasicDescription));
AudioUnitSetProperty(converter44To8,kAudioUnitProperty_StreamFormat,kAudioUnitScope_Output,0,&asbd8K,sizeof(AudioStreamBasicDescription));

AudioComponentInstance converter8To44;
AudioUnitSetProperty(converter8To44,kAudioUnitProperty_StreamFormat,kAudioUnitScope_Input,0,&asbd8K,sizeof(AudioStreamBasicDescription));
AudioUnitSetProperty(converter8To44,kAudioUnitProperty_StreamFormat,kAudioUnitScope_Output,0,& ioASBDout,sizeof(AudioStreamBasicDescription));

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