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parameter tuning

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*some filter parameter tinkering
* restore FFT_MajPeakSmth in UDP SoundSync receiver
This commit is contained in:
Frank
2024-07-05 15:12:41 +02:00
parent 160e66a766
commit cba883c663
1 changed files with 12 additions and 12 deletions
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24
usermods/audioreactive/audio_reactive.h
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@@ -240,7 +240,7 @@ const double agcFollowSlow[AGC_NUM_PRESETS] = {1/6144.f,1/4096.f,1/8192.f}; //
const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter
const double agcControlKi[AGC_NUM_PRESETS] = { 1.7f, 1.85f, 1.2f}; // AGC - PI control, integral gain parameter const double agcControlKi[AGC_NUM_PRESETS] = { 1.7f, 1.85f, 1.2f}; // AGC - PI control, integral gain parameter
#if defined(WLEDMM_FASTPATH) #if defined(WLEDMM_FASTPATH)
const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/6.f, 1/5.f, 1/10.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value) const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/8.f, 1/5.f, 1/12.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value)
#else #else
const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value) const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value)
#endif #endif
@@ -1434,13 +1434,8 @@ class AudioReactive : public Usermod {
{ {
float sampleAdj; // Gain adjusted sample value float sampleAdj; // Gain adjusted sample value
float tmpSample; // An interim sample variable used for calculations. float tmpSample; // An interim sample variable used for calculations.
#ifdef WLEDMM_FASTPATH const float weighting = 0.18f; // Exponential filter weighting. Will be adjustable in a future release.
constexpr float weighting = 0.35f; // slightly reduced filter strength, to reduce audio latency
constexpr float weighting2 = 0.25f;
#else
const float weighting = 0.2f; // Exponential filter weighting. Will be adjustable in a future release.
const float weighting2 = 0.073f; // Exponential filter weighting, for rising signal (a bit more robust against spikes) const float weighting2 = 0.073f; // Exponential filter weighting, for rising signal (a bit more robust against spikes)
#endif
const int AGC_preset = (soundAgc > 0)? (soundAgc-1): 0; // make sure the _compiler_ knows this value will not change while we are inside the function const int AGC_preset = (soundAgc > 0)? (soundAgc-1): 0; // make sure the _compiler_ knows this value will not change while we are inside the function
static bool isFrozen = false; static bool isFrozen = false;
static bool haveSilence = true; static bool haveSilence = true;
@@ -1492,9 +1487,11 @@ class AudioReactive : public Usermod {
if ((micLevelMethod == 2) && !haveSilence && (expAdjF >= (1.5f * float(soundSquelch)))) if ((micLevelMethod == 2) && !haveSilence && (expAdjF >= (1.5f * float(soundSquelch))))
isFrozen = true; // fast freeze mode: freeze micLevel once the volume rises 50% above squelch isFrozen = true; // fast freeze mode: freeze micLevel once the volume rises 50% above squelch
//expAdjF = (micInNoDC <= soundSquelch) ? 0: expAdjF; // simple noise gate - experimental // simple noise gate
expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate if ((expAdjF <= soundSquelch) || ((soundSquelch == 0) && (expAdjF < 0.25f))) {
if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0" expAdjF = 0.0f;
micInNoDC = 0.0f;
}
if (expAdjF <= 0.5f) if (expAdjF <= 0.5f)
haveSilence = true; haveSilence = true;
@@ -1515,7 +1512,7 @@ class AudioReactive : public Usermod {
sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // Adjust the gain. with inputLevel adjustment sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // Adjust the gain. with inputLevel adjustment
sampleReal = tmpSample; sampleReal = tmpSample;
sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ??? sampleAdj = fmax(fmin(sampleAdj, 255.0f), 0.0f); // Question: why are we limiting the value to 8 bits ???
sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!! sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!!
// keep "peak" sample, but decay value if current sample is below peak // keep "peak" sample, but decay value if current sample is below peak
@@ -1538,7 +1535,7 @@ class AudioReactive : public Usermod {
if (sampleMax < 0.5f) sampleMax = 0.0f; if (sampleMax < 0.5f) sampleMax = 0.0f;
#if defined(WLEDMM_FASTPATH) #if defined(WLEDMM_FASTPATH)
sampleAvg = ((sampleAvg * 7.0f) + sampleAdj) / 8.0f; // make reactions a bit more "crisp" in fastpath mode sampleAvg = ((sampleAvg * 11.0f) + sampleAdj) / 12.0f; // make reactions a bit more "crisp" in fastpath mode
#else #else
sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples. sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
#endif #endif
@@ -1791,6 +1788,9 @@ class AudioReactive : public Usermod {
my_magnitude = fmaxf(receivedPacket.FFT_Magnitude, 0.0f); my_magnitude = fmaxf(receivedPacket.FFT_Magnitude, 0.0f);
FFT_Magnitude = my_magnitude; FFT_Magnitude = my_magnitude;
FFT_MajorPeak = constrain(receivedPacket.FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects FFT_MajorPeak = constrain(receivedPacket.FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
#ifdef ARDUINO_ARCH_ESP32
FFT_MajPeakSmth = FFT_MajPeakSmth + 0.42f * (FFT_MajorPeak - FFT_MajPeakSmth); // simulate smooth value
#endif
agcSensitivity = 128.0f; // substitute - V2 format does not include this value agcSensitivity = 128.0f; // substitute - V2 format does not include this value
zeroCrossingCount = receivedPacket.zeroCrossingCount; zeroCrossingCount = receivedPacket.zeroCrossingCount;