我一直在尝试使用Apple的加速框架中的FFT来获取精确的频率,但我在弄清楚为什么我的值超出真实频率时遇到了麻烦。
我一直在使用这篇文章http://www.dspdimension.com/admin/pitch-shifting-using-the-ft/作为我实施的基础,经过努力达到我现在的目标之后,我完全被难倒了。
到目前为止,我已收到音频 - >汉宁窗口 - > FFT - >相位计算 - >奇怪的最终输出。我觉得某个地方的数学会有问题,但我现在真的没有想法了。
输出比应该低很多,例如,我输入440Hz并输出190Hz,或输入880Hz,输出400Hz。在大多数情况下,这些结果是一致的,但并非总是如此,并且似乎并不是任何事物之间的任何共同因素......
这是我的代码:
enum
{
sampleRate = 44100,
osamp = 4,
samples = 4096,
range = samples * 7 / 16,
step = samples / osamp
};
NSMutableArray *fftResults;
static COMPLEX_SPLIT A;
static FFTSetup setupReal;
static uint32_t log2n, n, nOver2;
static int32_t stride;
static float expct = 2*M_PI*((double)step/(double)samples);
static float phase1[range];
static float phase2[range];
static float dPhase[range];
- (void)fftSetup
{
// Declaring integers
log2n = 12;
n = 1 << log2n;
stride = 1;
nOver2 = n / 2;
// Allocating memory for complex vectors
A.realp = (float *) malloc(nOver2 * sizeof(float));
A.imagp = (float *) malloc(nOver2 * sizeof(float));
// Allocating memory for FFT
setupReal = vDSP_create_fftsetup(log2n, FFT_RADIX2);
// Setting phase
memset(phase2, 0, range * sizeof(float));
}
// For each sample in buffer...
for (int bufferCount = 0; bufferCount < audioBufferList.mNumberBuffers; bufferCount++)
{
// Declaring samples from audio buffer list
SInt16 *samples = (SInt16*)audioBufferList.mBuffers[bufferCount].mData;
// Creating Hann window function
for (int i = 0; i < nOver2; i++)
{
double hannMultiplier = 0.5 * (1 - cos((2 * M_PI * i) / (nOver2 - 1)));
// Applying window to each sample
A.realp[i] = hannMultiplier * samples[i];
A.imagp[i] = 0;
}
// Applying FFT
vDSP_fft_zrip(setupReal, &A, stride, log2n, FFT_FORWARD);
// Detecting phase
vDSP_zvphas(&A, stride, phase1, stride, range);
// Calculating phase difference
vDSP_vsub(phase2, stride, phase1, stride, dPhase, stride, range);
// Saving phase
memcpy(phase2, phase1, range * sizeof(float));
// Extracting DSP outputs
for (size_t j = 0; j < nOver2; j++)
{
NSNumber *realNumbers = [NSNumber numberWithFloat:A.realp[j]];
NSNumber *imagNumbers = [NSNumber numberWithFloat:A.imagp[j]];
[real addObject:realNumbers];
[imag addObject:imagNumbers];
}
// Combining real and imaginary parts
[resultsCombined addObject:real];
[resultsCombined addObject:imag];
// Filling FFT output array
[fftResults addObject:resultsCombined];
}
}
int fftCount = [fftResults count];
NSLog(@"FFT Count: %d",fftCount);
// For each FFT...
for (int i = 0; i < fftCount; i++)
{
// Declaring integers for peak detection
float peak = 0;
float binNumber = 0;
// Declaring integers for phase detection
float deltaPhase;
static float trueFrequency[range];
for (size_t j = 1; j < range; j++)
{
// Calculating bin magnitiude
float realVal = [[[[fftResults objectAtIndex:i] objectAtIndex:0] objectAtIndex:j] floatValue];
float imagVal = [[[[fftResults objectAtIndex:i] objectAtIndex:1] objectAtIndex:j] floatValue];
float magnitude = sqrtf(realVal*realVal + imagVal*imagVal);
// Peak detection
if (magnitude > peak)
{
peak = magnitude;
binNumber = (float)j;
}
// Getting phase difference
deltaPhase = dPhase[j];
// Subtract expected difference
deltaPhase -= (float)j*expct;
// Map phase difference into +/- pi interval
int qpd = deltaPhase / M_PI;
if (qpd >= 0)
qpd += qpd&1;
else
qpd -= qpd&1;
deltaPhase -= M_PI * (float)qpd;
// Getting bin deviation from +/i interval
float deltaFrequency = osamp * deltaPhase / (2 * M_PI);
// Calculating true frequency at the j-th partial
trueFrequency[j] = (j * (sampleRate/samples)) + (deltaFrequency * (sampleRate/samples));
}
UInt32 mag;
mag = binNumber;
// Extracting frequency at bin peak
float f = trueFrequency[mag];
NSLog(@"True frequency = %fHz", f);
float b = roundf(binNumber*(sampleRate/nOver2));
NSLog(@" Bin frequency = %fHz", b);
}
答案 0 :(得分:0)
请注意,预期的相位差(即使对于以bin为中心的频率)取决于FFT对的窗口偏移或重叠,以及FFT结果的bin编号或频率。例如如果您将窗口偏移很少(1个样本),则2个FFT之间的展开相位差将小于具有较大偏移的相位差。在相同的偏移处,如果频率较高,则两个FFT的相同二进制位之间的预期相位差将更大(或者它将包含更多)。