iOS – 生成并播放无限简单的音频(正弦波)

这是一个简单的应用程序,它将按需生成频率.你还没有指定是做iOS还是OSX,所以我已经选择了OSX,因为它稍微简单一些(没有搞乱音频会话类别).如果你需要iOS,你可以通过查看音频会话类别基础知识并交换RemoteIO音频单元的默认输出音频单元来找出丢失的位.

请注意,这样做的目的纯粹是为了演示一些核心音频/音频单元的基础知识.如果你想开始变得比这更复杂,你可能想要查看AUGraph API(也是为了提供一个干净的例子,我没有做任何错误检查.在处理Core Audio时总是要进行错误检查).

您需要将AudioToolBox和AudioUnit框架添加到项目中才能使用此代码.

#import <AudioToolBox/AudioToolBox.h>

@interface SWAppDelegate : NSObject <NSApplicationDelegate>
{
    AudioUnit outputUnit;
    double renderPhase;
}
@end

@implementation SWAppDelegate

- (void)applicationDidFinishLaunching:(NSNotification *)aNotification
{
//  First,we need to establish which Audio Unit we want.

//  We start with its description,which is:
    AudioComponentDescription outputUnitDescription = {
        .componentType         = kAudioUnitType_Output,.componentSubType      = kAudioUnitSubType_DefaultOutput,.componentManufacturer = kAudioUnitManufacturer_Apple
    };

//  Next,we get the first (and only) component corresponding to that description
    AudioComponent outputComponent = AudioComponentFindNext(NULL,&outputUnitDescription);

//  Now we can create an instance of that component,which will create an
//  instance of the Audio Unit we're looking for (the default output)
    AudioComponentInstanceNew(outputComponent,&outputUnit);
    AudioUnitinitialize(outputUnit);

//  Next we'll tell the output unit what format our generated audio will
//  be in. Generally speaking,you'll want to stick to sane formats,since
//  the output unit won't accept every single possible stream format.
//  Here,we're specifying floating point samples with a sample rate of
//  44100 Hz in mono (i.e. 1 channel)
    AudioStreamBasicDescription ASBD = {
        .mSampleRate       = 44100,.mFormatID         = kAudioFormatLinearPCM,.mFormatFlags      = kAudioFormatFlagsNativeFloatPacked,.mChannelsPerFrame = 1,.mFramesPerPacket  = 1,.mBitsPerChannel   = sizeof(Float32) * 8,.mBytesPerPacket   = sizeof(Float32),.mBytesPerFrame    = sizeof(Float32)
    };

    AudioUnitSetProperty(outputUnit,kAudioUnitProperty_StreamFormat,kAudioUnitScope_Input,&ASBD,sizeof(ASBD));

//  Next step is to tell our output unit which function we'd like it
//  to call to get audio samples. We'll also pass in a context pointer,//  which can be a pointer to anything you need to maintain state between
//  render callbacks. We only need to point to a double which represents
//  the current phase of the sine wave we're creating.
    AURenderCallbackStruct callbackInfo = {
        .inputProc       = SineWaveRenderCallback,.inputProcRefCon = &renderPhase
    };

    AudioUnitSetProperty(outputUnit,kAudioUnitProperty_SetRenderCallback,kAudioUnitScope_Global,&callbackInfo,sizeof(callbackInfo));

//  Here we're telling the output unit to start requesting audio samples
//  from our render callback. This is the line of code that starts actually
//  sending audio to your speakers.
    AudioOutputUnitStart(outputUnit);
}

// This is our render callback. It will be called very frequently for short
// buffers of audio (512 samples per call on my machine).
Osstatus SineWaveRenderCallback(void * inRefCon,AudioUnitRenderActionFlags * ioActionFlags,const AudioTimeStamp * inTimeStamp,UInt32 inBusNumber,UInt32 inNumberFrames,audiobufferlist * ioData)
{
    // inRefCon is the context pointer we passed in earlier when setting the render callback
    double currentPhase = *((double *)inRefCon);
    // ioData is where we're supposed to put the audio samples we've created
    Float32 * outputBuffer = (Float32 *)ioData->mBuffers[0].mData;
    const double frequency = 440.;
    const double phaseStep = (frequency / 44100.) * (M_PI * 2.);

    for(int i = 0; i < inNumberFrames; i++) {
        outputBuffer[i] = sin(currentPhase);
        currentPhase += phaseStep;
    }

    // If we were doing stereo (or more),this would copy our sine wave samples
    // to all of the remaining channels
    for(int i = 1; i < ioData->mNumberBuffers; i++) {
        memcpy(ioData->mBuffers[i].mData,outputBuffer,ioData->mBuffers[i].mDataByteSize);
    }

    // writing the current phase back to inRefCon so we can use it on the next call
    *((double *)inRefCon) = currentPhase;
    return noErr;
}

- (void)applicationWillTerminate:(NSNotification *)notification
{
    AudioOutputUnitStop(outputUnit);
    AudioUnitUninitialize(outputUnit);
    AudioComponentInstancedispose(outputUnit);
}

@end

您可以随意调用AudioOutputUnitStart()和AudioOutputUnitStop()来开始/停止生成音频.如果要动态更改频率,可以传入指向包含renderPhase double和另一个表示所需频率的struct的指针.

在渲染回调中要小心.它是从实时线程调用的(不是来自与主运行循环相同的线程).渲染回调受到一些相当严格的时间要求的限制,这意味着你的回调中不应该做很多事情,例如：

>分配内存
>等待互斥锁
>从磁盘上的文件中读取
> Objective-C消息传递(是的,认真的.)

请注意,这不是执行此操作的唯一方法.我只是用这种方式演示了它,因为你已经标记了这个核心音频.如果您不需要更改频率,可以将AVAudioPlayer与包含正弦波的预制声音文件一起使用.

还有Novocaine,它隐藏了很多这种冗长.您还可以查看Audio Queue API,它与我编写的Core Audio示例非常相似,但是您可以将它与硬件分离得更多(即,它对您在渲染回调中的行为方式不太严格).