From 17e6a887cb02a9b9f5a3ddeffc72aa6a2118b8c3 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Wed, 23 Nov 2022 14:39:19 +0100 Subject: [PATCH] FFT code restructuring, part2 puting mic filter and FFT post-processing into seperate funtions, to improve code clarity. --- usermods/audioreactive/audio_reactive.h | 122 +++++++++++++----------- 1 file changed, 64 insertions(+), 58 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index ac2673cf..6e29d22d 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -303,35 +303,7 @@ void FFTcode(void * parameter) // band pass filter - can reduce noise floor by a factor of 50 // downside: frequencies below 100Hz will be ignored - if (useBandPassFilter) { - // 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; - } - } + if (useBandPassFilter) runMicFilter(samplesFFT, vReal); // find highest sample in the 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) 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 (fabsf(volumeSmth) > 0.25f) { // noise gate open +#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 // +/////////////////////////// + +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. fftCalc[i] *= fftResultPink[pinkIndex][i]; 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 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 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]; @@ -543,31 +573,7 @@ void FFTcode(void * parameter) } 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 // ////////////////////