Question

我正在设置我的音频输入和音频输出流，如下所示。问题是无论我如何设置格式和缓冲区大小，A2DP的输出延迟总是在0.15s左右。我已经发布了以下配置。这里outLat是测量的输出延迟。

如何在不牺牲音质的情况下进一步降低音频输出延迟？

- (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;

    // 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;
}

Answer 1

在AVAudioSession中使用Measurement Mode，并了解其目的类别及其限制。

AVAudio Session Mode Measurement

具有A2DP输出通道的AVAudioSession的延迟延迟

1 个答案: