核心音频,Goertzel算法不起作用

123*_*321 5 iphone audio signal-processing objective-c core-audio

我目前正在创建一个应用程序,它可以从iPhone的麦克风实时计算出预定频率(16780Hz)的幅度.

我将声音数据放在缓冲区中,然后尝试使用Goertzel处理它,Goertzel是为此任务设计的算法.Goertzel信息.这是问题的开始.

当记录的声音比定义的声音(167Hz)低得多(5000Hz)时,该算法以非常积极的结果作出响应.事实上,结果远比记录正确频率的声音时产生的结果更为积极.

这是我对goertzel的实现:

double goertzel(unsigned short *sample, int sampleRate, double Freq, int len )
{

double realW = 2.0 * cos(2.0 * M_PI * Freq / sampleRate);
double imagW = 2.0 * sin(2.0 * M_PI * Freq / sampleRate);
double d1 = 0;
double d2 = 0;
int z;
double y;
for (int i = 0; i < len; i++) {
    y=(double)(signed short)sample[i] +realW * d1 - d2;
    d2 = d1;
    d1 = y;
}
double rR = 0.5 * realW *d1-d2;
double rI = 0.5 * imagW *d1-d2;

return (sqrt(pow(rR, 2)+pow(rI,2)))/len;
} /* end function goertzel */
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以下是我检索音频的方法,如果它完全相关的话

-(void)startListeningWithFrequency:(float)frequency;
{
OSStatus status;
//AudioComponentInstance audioUnit;
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;

AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
status = AudioComponentInstanceNew( inputComponent, &audioUnit);
checkStatus(status);

UInt32 flag = 1;
status = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input,kInputBus, &flag, sizeof(flag));
checkStatus(status);

AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate         = 44100.00;//44100.00;
audioFormat.mFormatID           = kAudioFormatLinearPCM;
audioFormat.mFormatFlags        = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;
audioFormat.mFramesPerPacket    = 1;
audioFormat.mChannelsPerFrame   = 1;
audioFormat.mBitsPerChannel     = 16;
//  float
audioFormat.mBytesPerPacket     = 2;
audioFormat.mBytesPerFrame      = 2;

status = AudioUnitSetProperty(audioUnit,
                              kAudioUnitProperty_StreamFormat,
                              kAudioUnitScope_Output,
                              kInputBus,
                              &audioFormat, 
                              sizeof(audioFormat));
checkStatus(status);
//status = AudioUnitSetProperty(audioUnit, 
//                            kAudioUnitProperty_StreamFormat, 
//                            kAudioUnitScope_Input, 
//                            kOutputBus, 
//                            &audioFormat, 
//                            sizeof(audioFormat));
checkStatus(status);
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = self;
status = AudioUnitSetProperty(audioUnit, 
                              kAudioOutputUnitProperty_SetInputCallback,
                              kAudioUnitScope_Global,
                              kInputBus, &callbackStruct, sizeof(callbackStruct));
checkStatus(status);
/*  UInt32 shouldAllocateBuffer = 1;
AudioUnitSetProperty(audioUnit, kAudioUnitProperty_ShouldAllocateBuffer, kAudioUnitScope_Global, 1, &shouldAllocateBuffer, sizeof(shouldAllocateBuffer));
*/
status = AudioOutputUnitStart(audioUnit);

}
static OSStatus recordingCallback(void *inRefCon, 
                              AudioUnitRenderActionFlags *ioActionFlags, 
                              const AudioTimeStamp *inTimeStamp, 
                              UInt32 inBusNumber, 
                              UInt32 inNumberFrames, 
                              AudioBufferList *ioData) {
AudioBuffer buffer;

buffer.mNumberChannels = 1;
buffer.mDataByteSize = inNumberFrames * 2;
//NSLog(@"%d",inNumberFrames);
buffer.mData = malloc( inNumberFrames * 2 );

// Put buffer in a AudioBufferList
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = buffer;



OSStatus status;
status = AudioUnitRender(audioUnit, 
                         ioActionFlags, 
                         inTimeStamp, 
                         inBusNumber, 
                         inNumberFrames, 
                         &bufferList);  
checkStatus(status);
//double g = calculateGoertzel((const char *)(&bufferList)->mBuffers[0].mData,16789.0,96000.0);
UInt16 *q = (UInt16 *)(&bufferList)->mBuffers[0].mData;
int N = sizeof(q)/sizeof(UInt16);
double Qr,Qi;
double theta = 2.0*M_PI*16780/44100;
double g = goertzel(q,44100,16780,N);

NSLog(@"goertzel:%f", g);
}
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这将返回数百个数字,频率远低于16780Hz,而对于16780Hz的频率,返回的数字要小得多.

我非常沮丧,非常感谢帮助.

Vag*_*ant 3

只是一个猜测:

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根据 Nyquist\xe2\x80\x93Shannon 采样定理,采样率应至少是您尝试测量的频率的两倍。而你的是,但只是勉强。44.1kHz 的采样率是测量 22kHz 信号的外沿。16kHz 的信号足够接近混叠可能导致波形分析出现问题的极限。这是一张图片来说明我的观点:\n在此输入图像描述

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所以,我猜你需要更高的采样率。为什么不尝试通过该算法运行纯 16kHz 正弦波,看看它是否能做得更好?如果测试数据中只有单一频率,则混叠问题就不那么严重了。如果您从正弦波中获得更高的响应,那么您可能只需要更高的采样率。

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