从内存数据流在iOS上播放音频

我正在将一个音频库移植到iOS，允许播放回调反馈的音频流。用户提供了一个返回原始PCM数据的回调，我需要播放这些数据。此外，库必须能够同时播放多个流

我想我需要使用AVFoundation，但AVAudioPlayer似乎不支持流式音频缓冲区，而且我可以找到的所有流式文档都直接来自网络。我应该在这里使用什么API

提前谢谢

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顺便说一句，我不是通过Swift或Objective-C使用苹果库。但是我假设所有的东西都是公开的，所以无论如何，Swift中的一个例子将非常感谢

您需要初始化：

音频会话使用输入音频单元和输出

-(SInt32) audioSessionInitialization:(SInt32)preferred_sample_rate {

    // - - - - - - Audio Session initialization
    NSError *audioSessionError = nil;
    session = [AVAudioSession sharedInstance];

    // disable AVAudioSession
    [session setActive:NO error:&audioSessionError];

    // set category - (PlayAndRecord to use input and output session AudioUnits)
   [session setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:AVAudioSessionCategoryOptionDefaultToSpeaker error:&audioSessionError];

   double preferredSampleRate = 441000;
   [session setPreferredSampleRate:preferredSampleRate error:&audioSessionError];

   // enable AVAudioSession
   [session setActive:YES error:&audioSessionError];


   // Configure notification for device output change (speakers/headphones)
   [[NSNotificationCenter defaultCenter] addObserver:self
                                     selector:@selector(routeChange:)
                                         name:AVAudioSessionRouteChangeNotification
                                       object:nil];


   // - - - - - - Create audio engine
   [self audioEngineInitialization];

   return [session sampleRate];
 }

音频引擎

-(void) audioEngineInitialization{

    engine = [[AVAudioEngine alloc] init];
    inputNode = [engine inputNode];
    outputNode = [engine outputNode];

    [engine connect:inputNode to:outputNode format:[inputNode inputFormatForBus:0]];


    AudioStreamBasicDescription asbd_player;
    asbd_player.mSampleRate        = session.sampleRate;
    asbd_player.mFormatID            = kAudioFormatLinearPCM;
    asbd_player.mFormatFlags        = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
    asbd_player.mFramesPerPacket    = 1;
    asbd_player.mChannelsPerFrame    = 2;
    asbd_player.mBitsPerChannel    = 16;
    asbd_player.mBytesPerPacket    = 4;
    asbd_player.mBytesPerFrame        = 4;

    OSStatus status;
    status = AudioUnitSetProperty(inputNode.audioUnit,
                              kAudioUnitProperty_StreamFormat,
                              kAudioUnitScope_Input,
                              0,
                              &asbd_player,
                              sizeof(asbd_player));


    // Add the render callback for the ioUnit: for playing
    AURenderCallbackStruct callbackStruct;
    callbackStruct.inputProc = engineInputCallback; ///CALLBACK///
    callbackStruct.inputProcRefCon = (__bridge void *)(self);
    status = AudioUnitSetProperty(inputNode.audioUnit,
                              kAudioUnitProperty_SetRenderCallback,
                              kAudioUnitScope_Input,//Global
                              kOutputBus,
                              &callbackStruct,
                              sizeof(callbackStruct));

    [engine prepare];
}

音频引擎回调

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

    // the reference to the audio controller where you get the stream data
    MyAudioController *ac = (__bridge MyAudioController *)(inRefCon);

    // in practice we will only ever have 1 buffer, since audio format is mono
    for (int i = 0; i < ioData->mNumberBuffers; i++) { 
        AudioBuffer buffer = ioData->mBuffers[i];

        // copy stream buffer data to output buffer
        UInt32 size = min(buffer.mDataByteSize, ac.playbackBuffer.mDataByteSize); 
        memcpy(buffer.mData, ac.streamBuffer.mData, size);
        buffer.mDataByteSize = size; // indicate how much data we wrote in the buffer
    }

    return noErr;
}

静态OSStatus engineInputCallback（在refcon中无效*，
AudioUnitRenderActionFlags*ioActionFlags，
常量音频时间戳*inTimeStamp，
UInt32 InBunsNumber，
UInt32数字帧，
音频缓冲列表*ioData）
{
//对获取流数据的音频控制器的引用
MyAudioController*ac=（桥接MyAudioController*）（inRefCon）；
//实际上我们只有一个缓冲区，因为音频格式是单声道的
对于（inti=0；imNumberBuffers；i++）{
AudioBuffer=ioData->mBuffers[i]；
//将流缓冲区数据复制到输出缓冲区
UInt32 size=min（buffer.mDataByteSize，ac.playbackuffer.mDataByteSize）；
memcpy（buffer.mData、ac.streamBuffer.mData、size）；
buffer.mDataByteSize=size；//指示我们在缓冲区中写入了多少数据
}
返回noErr；
}

---

AVAudioEngine

+

AVAudioPlayerNode

让我们从缓冲区播放音频。下面是一个使用单个播放器的示例，其中音频缓冲区取自麦克风：谢谢大家，音频队列正是我所需要的！