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我正在寻找一个非常简单的 iOS 应用程序，它带有一个可以启动和停止音频信号的按钮。信号只是一个正弦波，它会在整个播​​放过程中检查我的模型（音量的实例变量）并相应地改变它的音量。

我的困难与任务的不确定性有关。我了解如何构建表格、用数据填充表格、响应按钮按下等等；但是，当涉及到无限期地继续进行某些事情时（在这种情况下，是声音），我有点卡住了！任何指针都会很棒！

谢谢阅读。

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1 回答 1

17

这是一个简单的应用程序，它将按需播放生成的频率。你还没有指定是做 iOS 还是 OSX，所以我选择了 OSX，因为它稍微简单一些（不会弄乱音频会话类别）。如果您需要 iOS，您可以通过查看音频会话类别基础知识并将默认输出音频单元替换为 RemoteIO 音频单元来找出缺失的部分。

请注意，这样做的目的纯粹是为了演示一些 Core Audio / Audio Unit 基础知识。如果您想开始变得比这更复杂，您可能需要查看AUGraphAPI（也为了提供一个干净的示例，我没有进行任何错误检查。在处理 Core Audio 时始终进行错误检查） .

您需要将AudioToolbox和AudioUnit框架添加到您的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,
                         0,
                         &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,
                         0,
                         &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

您可以随意拨打电话AudioOutputUnitStart()开始AudioOutputUnitStop()/停止制作音频。如果你想动态改变频率，你可以传入一个指针，指向一个struct既包含renderPhase double 又包含另一个代表你想要的频率的指针。

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，这意味着在你的回调中有很多你不应该做的事情，例如：

  • 分配内存
  • 等待互斥锁
  • 从磁盘上的文件读取
  • 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。自从您标记了此核心音频后，我才以这种方式进行了演示。如果您不需要更改频率，您可以使用AVAudioPlayer包含您的正弦波的预制声音文件。

还有Novocaine，它对你隐藏了很多这种冗长的内容。您还可以查看 Audio Queue API，它的工作方式与我编写的 Core Audio 示例非常相似，但将您与硬件分离了一点（即，它对您在渲染回调中的行为方式不那么严格）。

于 2013-01-23T11:18:10.997 回答