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FFT code restructuring, part2

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puting mic filter and FFT post-processing into seperate funtions, to improve code clarity.
This commit is contained in:
Frank
2022-11-23 14:39:19 +01:00
parent 068e88453c
commit 17e6a887cb
1 changed files with 64 additions and 58 deletions
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122
usermods/audioreactive/audio_reactive.h
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@@ -303,35 +303,7 @@ void FFTcode(void * parameter)
// band pass filter - can reduce noise floor by a factor of 50 // band pass filter - can reduce noise floor by a factor of 50
// downside: frequencies below 100Hz will be ignored // downside: frequencies below 100Hz will be ignored
if (useBandPassFilter) { if (useBandPassFilter) runMicFilter(samplesFFT, vReal);
// low frequency cutoff parameter - see https://dsp.stackexchange.com/questions/40462/exponential-moving-average-cut-off-frequency
//constexpr float alpha = 0.04f; // 150Hz
//constexpr float alpha = 0.03f; // 110Hz
constexpr float alpha = 0.0225f; // 80hz
//constexpr float alpha = 0.01693f;// 60hz
// high frequency cutoff parameter
//constexpr float beta1 = 0.75; // 11Khz
//constexpr float beta1 = 0.82; // 15Khz
//constexpr float beta1 = 0.8285; // 18Khz
constexpr float beta1 = 0.85; // 20Khz
constexpr float beta2 = (1.0f - beta1) / 2.0;
static float last_vals[2] = { 0.0f }; // FIR high freq cutoff filter
static float lowfilt = 0.0f; // IIR low frequency cutoff filter
for (int i=0; i < samplesFFT; i++) {
// FIR lowpass, to remove high frequency noise
float highFilteredSample;
if (i < (samplesFFT-1)) highFilteredSample = beta1*vReal[i] + beta2*last_vals[0] + beta2*vReal[i+1]; // smooth out spikes
else highFilteredSample = beta1*vReal[i] + beta2*last_vals[0] + beta2*last_vals[1]; // spcial handling for last sample in array
last_vals[1] = last_vals[0];
last_vals[0] = vReal[i];
vReal[i] = highFilteredSample;
// IIR highpass, to remove low frequency noise
lowfilt += alpha * (vReal[i] - lowfilt);
vReal[i] = vReal[i] - lowfilt;
}
}
// find highest sample in the batch // find highest sample in the batch
float maxSample = 0.0f; // max sample from FFT batch float maxSample = 0.0f; // max sample from FFT batch
@@ -465,10 +437,68 @@ void FFTcode(void * parameter)
// post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling) // post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
if (pinkIndex > MAX_PINK) pinkIndex = MAX_PINK; if (pinkIndex > MAX_PINK) pinkIndex = MAX_PINK;
for (int i=0; i < NUM_GEQ_CHANNELS; i++) { //postProcessFFTResults((fabsf(sampleAvg) > 0.25f)? true : false , NUM_GEQ_CHANNELS);
postProcessFFTResults((fabsf(volumeSmth)>0.25f)? true : false , NUM_GEQ_CHANNELS);
//if (fabsf(sampleAvg) > 0.25f) { // noise gate open #if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS)
if (fabsf(volumeSmth) > 0.25f) { // noise gate open if (haveDoneFFT && (start < esp_timer_get_time())) { // filter out overflows
uint64_t fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding
fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth
}
#endif
// run peak detection
autoResetPeak();
detectSamplePeak();
#if !defined(I2S_GRAB_ADC1_COMPLETELY)
if ((audioSource == nullptr) || (audioSource->getType() != AudioSource::Type_I2SAdc)) // the "delay trick" does not help for analog ADC
#endif
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
} // for(;;)ever
} // FFTcode() task end
///////////////////////////
// Pre / Postprocessing //
///////////////////////////
static void runMicFilter(uint16_t numSamples, float *sampleBuffer) // pre-filtering of raw samples (band-pass)
{
// low frequency cutoff parameter - see https://dsp.stackexchange.com/questions/40462/exponential-moving-average-cut-off-frequency
//constexpr float alpha = 0.04f; // 150Hz
//constexpr float alpha = 0.03f; // 110Hz
constexpr float alpha = 0.0225f; // 80hz
//constexpr float alpha = 0.01693f;// 60hz
// high frequency cutoff parameter
//constexpr float beta1 = 0.75; // 11Khz
//constexpr float beta1 = 0.82; // 15Khz
//constexpr float beta1 = 0.8285; // 18Khz
constexpr float beta1 = 0.85; // 20Khz
constexpr float beta2 = (1.0f - beta1) / 2.0;
static float last_vals[2] = { 0.0f }; // FIR high freq cutoff filter
static float lowfilt = 0.0f; // IIR low frequency cutoff filter
for (int i=0; i < numSamples; i++) {
// FIR lowpass, to remove high frequency noise
float highFilteredSample;
if (i < (numSamples-1)) highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*sampleBuffer[i+1]; // smooth out spikes
else highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*last_vals[1]; // spcial handling for last sample in array
last_vals[1] = last_vals[0];
last_vals[0] = sampleBuffer[i];
sampleBuffer[i] = highFilteredSample;
// IIR highpass, to remove low frequency noise
lowfilt += alpha * (sampleBuffer[i] - lowfilt);
sampleBuffer[i] = sampleBuffer[i] - lowfilt;
}
}
static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels) // post-processing and post-amp of GEQ channels
{
for (int i=0; i < numberOfChannels; i++) {
if (noiseGateOpen) { // noise gate open
// Adjustment for frequency curves. // Adjustment for frequency curves.
fftCalc[i] *= fftResultPink[pinkIndex][i]; fftCalc[i] *= fftResultPink[pinkIndex][i];
if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function
@@ -479,7 +509,7 @@ void FFTcode(void * parameter)
// smooth results - rise fast, fall slower // smooth results - rise fast, fall slower
if(fftCalc[i] > fftAvg[i]) // rise fast if(fftCalc[i] > fftAvg[i]) // rise fast
fftAvg[i] = fftCalc[i] *0.75f + 0.25f*fftAvg[i]; // will need approx 2 cycles (50ms) for converging against fftCalc[i] fftAvg[i] = fftCalc[i] *0.78f + 0.22f*fftAvg[i]; // will need approx 1-2 cycles (50ms) for converging against fftCalc[i]
else { // fall slow else { // fall slow
if (decayTime < 250) fftAvg[i] = fftCalc[i]*0.4f + 0.6f*fftAvg[i]; if (decayTime < 250) fftAvg[i] = fftCalc[i]*0.4f + 0.6f*fftAvg[i];
else if (decayTime < 500) fftAvg[i] = fftCalc[i]*0.33f + 0.67f*fftAvg[i]; else if (decayTime < 500) fftAvg[i] = fftCalc[i]*0.33f + 0.67f*fftAvg[i];
@@ -543,31 +573,7 @@ void FFTcode(void * parameter)
} }
fftResult[i] = constrain((int)currentResult, 0, 255); fftResult[i] = constrain((int)currentResult, 0, 255);
} }
}
#if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS)
if (haveDoneFFT && (start < esp_timer_get_time())) { // filter out overflows
uint64_t fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding
fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth
}
#endif
// run peak detection
autoResetPeak();
detectSamplePeak();
#if !defined(I2S_GRAB_ADC1_COMPLETELY)
if ((audioSource == nullptr) || (audioSource->getType() != AudioSource::Type_I2SAdc)) // the "delay trick" does not help for analog ADC
#endif
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
} // for(;;)ever
} // FFTcode() task end
///////////////////////////
// Pre / Postprocessing //
///////////////////////////
//////////////////// ////////////////////
// Peak detection // // Peak detection //
//////////////////// ////////////////////