我的困难与任务的不确定性有关.我理解如何构建表格,填充数据,响应按钮按下等等;然而,当谈到只是无限期地继续(在这种情况下,声音)时,我有点卡住了!任何指针都会很棒!
谢谢阅读.
解决方法 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 *** 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).
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