iOS - 生成并播放不确定的简单音频(正弦波)

Question

我想为 iOS 构建一个非常简单的应用程序，它带有一个按钮来启动和停止音频信号。信号将只是一个正弦波，它将在整个播放过程中检查我的模型(音量的实例变量)并相应地更改其音量。

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

感谢阅读。

Answer 1

这是一个简单的应用程序，它将按需播放生成的频率。你没有指定是做 iOS 还是 OSX，所以我选择了 OSX，因为它稍微简单一些(不会弄乱 Audio Session 类别)。如果您需要 iOS，您将能够通过查看 Audio Session 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，这纯粹是为了演示一些核心音频/音频单元基础知识。如果您想开始变得比这更复杂，您可能需要查看 AUGraph API(同样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 时总是进行错误检查。

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

您可以随意调用AudioOutputUnitStart() 和AudioOutputUnitStop() 来开始/停止产生音频。如果您想动态更改频率，您可以传入一个指向 struct 的指针，其中包含 renderPhase double 和另一个表示您想要的频率的结构。

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)。渲染回调有一些相当严格的时间要求，这意味着您不应该在回调中做很多事情，例如:

• 分配内存
• 等待互斥锁
• 读取磁盘上的文件
• Objective-C 消息传递(是的，认真的。)

请注意，这不是唯一的方法。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，您可以将 AVAudioPlayer 与包含正弦波的预制声音文件一起使用。

还有 Novocaine ，这对你隐藏了很多这样的冗长。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，但将您与硬件分离更多一点(即，它对您在渲染回调中的行为不那么严格)。