iOS - 生成並播放無限期，簡單的音頻（正弦波）

Question

我正在尋找一個啓動和停止音頻信號的按鈕，爲iOS構建一個令人難以置信的簡單應用程序。信號只是一個正弦波，它將在整個回放過程中檢查我的模型（音量的一個實例變量）並相應地更改其音量。

我的困難與任務的不確定性有關。我瞭解如何構建表格，填充數據，響應按鈕等等;然而，當談到無限期地繼續下去（在這種情況下，聲音）時，我有點卡住了！任何指針都會很棒！

感謝您的閱讀。

Answer 1

這是一個簡單的應用程序，它將播放按需生成的頻率。你還沒有指定是否執行iOS或OSX，所以我已經選擇了OSX，因爲它稍微簡單一些（不會搞亂音頻會話類別）。如果您需要iOS，您可以通過查看音頻會話類別基礎知識並交換RemoteIO音頻單元的默認輸出音頻單元來找出缺失的位。

請注意，這樣做的目的純粹是爲了演示一些核心音頻/音頻單元的基礎知識。如果你想開始變得比這更復雜，你可能需要查看AUGraph API（爲了提供一個乾淨的例子，我沒有做任何錯誤檢查。總是在處理時做錯誤檢查覈心音頻）。

您需要將AudioToolbox和AudioUnit框架添加到您的項目才能使用此代碼。

#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 

您可以隨意調用AudioOutputUnitStart()和AudioOutputUnitStop()啓動/停止生產音頻。如果你想動態改變頻率，你可以傳遞一個指向struct的指針，該指針包含renderPhase double和另一個代表你想要的頻率。

請注意呈現回調。它是從實時線程調用的（不是與主運行循環相同的線程）。渲染回調受到一些相當嚴格的時間要求，這意味着有很多事情你不應該在回調做，如：

• 從文件分配內存
• 等待一個互斥
• 閱讀在磁盤上
• Objective-C的消息（是的，嚴重的。）

請注意，這是不是這樣做的唯一途徑。我只是用這種方式演示了它，因爲你已經標記了這個核心音頻。如果您不需要更改頻率，則只需使用帶有包含正弦波的預製聲音文件的AVAudioPlayer即可。

還有Novocaine，它隱藏了很多這種冗長的你。您還可以查看Audio Queue API，它與我編寫的Core Audio示例非常相似，但將其從硬件中分離出一點點（即對渲染回調的表現不太嚴格）。