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AR: removing some old debug code

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Align with SR WLED code:
- removed old debug code that did not work any more
- removed experimental MAJORPEAK_SUPPRESS_NOISE code
This commit is contained in:
Frank 2022-08-06 17:24:39 +02:00
parent 96d497a5cd
commit d0f53cb14a
1 changed files with 27 additions and 129 deletions
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156
usermods/audioreactive/audio_reactive.h
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@ -21,7 +21,13 @@
*/ */
// Comment/Uncomment to toggle usb serial debugging // Comment/Uncomment to toggle usb serial debugging
// #define SR_DEBUG // #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter)
// #define FFT_SAMPLING_LOG // FFT result debugging
// #define SR_DEBUG // generic SR DEBUG messages
// hackers corner
// nothing atm
#ifdef SR_DEBUG #ifdef SR_DEBUG
#define DEBUGSR_PRINT(x) Serial.print(x) #define DEBUGSR_PRINT(x) Serial.print(x)
#define DEBUGSR_PRINTLN(x) Serial.println(x) #define DEBUGSR_PRINTLN(x) Serial.println(x)
@ -31,6 +37,11 @@
#define DEBUGSR_PRINTLN(x) #define DEBUGSR_PRINTLN(x)
#define DEBUGSR_PRINTF(x...) #define DEBUGSR_PRINTF(x...)
#endif #endif
// legacy support
#if defined(SR_DEBUG) && !defined(MIC_LOGGER) && !defined(NO_MIC_LOGGER)
#define MIC_LOGGER
#endif
#include "audio_source.h" #include "audio_source.h"
@ -39,12 +50,6 @@ constexpr int BLOCK_SIZE = 128;
constexpr int SAMPLE_RATE = 20480; // Base sample rate in Hz - 20Khz is experimental constexpr int SAMPLE_RATE = 20480; // Base sample rate in Hz - 20Khz is experimental
//constexpr int SAMPLE_RATE = 10240; // Base sample rate in Hz - standard //constexpr int SAMPLE_RATE = 10240; // Base sample rate in Hz - standard
// #define MIC_LOGGER
// #define MIC_SAMPLING_LOG
// #define FFT_SAMPLING_LOG
//#define MAJORPEAK_SUPPRESS_NOISE // define to activate a dirty hack that ignores the lowest + hightest FFT bins
// globals // globals
static uint8_t inputLevel = 128; // UI slider value static uint8_t inputLevel = 128; // UI slider value
static uint8_t soundSquelch = 10; // squelch value for volume reactive routines (config value) static uint8_t soundSquelch = 10; // squelch value for volume reactive routines (config value)
@ -146,9 +151,6 @@ float fftAddAvg(int from, int to) {
void FFTcode(void * parameter) void FFTcode(void * parameter)
{ {
DEBUGSR_PRINT("FFT running on core: "); DEBUGSR_PRINTLN(xPortGetCoreID()); DEBUGSR_PRINT("FFT running on core: "); DEBUGSR_PRINTLN(xPortGetCoreID());
#ifdef MAJORPEAK_SUPPRESS_NOISE
static float xtemp[24] = {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0};
#endif
for(;;) { for(;;) {
delay(1); // DO NOT DELETE THIS LINE! It is needed to give the IDLE(0) task enough time and to keep the watchdog happy. delay(1); // DO NOT DELETE THIS LINE! It is needed to give the IDLE(0) task enough time and to keep the watchdog happy.
@ -161,24 +163,18 @@ void FFTcode(void * parameter)
} }
#ifdef WLED_DEBUG #ifdef WLED_DEBUG
// unsigned long start = millis();
uint64_t start = esp_timer_get_time(); uint64_t start = esp_timer_get_time();
#endif #endif
if (audioSource) audioSource->getSamples(vReal, samplesFFT); if (audioSource) audioSource->getSamples(vReal, samplesFFT);
#ifdef WLED_DEBUG #ifdef WLED_DEBUG
//sampleTime = ((millis() - start)*3 + sampleTime*7)/10; // smooth
if (start < esp_timer_get_time()) { // filter out overflows if (start < esp_timer_get_time()) { // filter out overflows
unsigned long sampleTimeInMillis = (esp_timer_get_time() - start +500ULL) / 1000ULL; // "+500" to ensure proper rounding unsigned long sampleTimeInMillis = (esp_timer_get_time() - start +500ULL) / 1000ULL; // "+500" to ensure proper rounding
sampleTime = (sampleTimeInMillis*3 + sampleTime*7)/10; // smooth sampleTime = (sampleTimeInMillis*3 + sampleTime*7)/10; // smooth
} }
#endif #endif
// old code - Last sample in vReal is our current mic sample
//micDataSm = (uint16_t)vReal[samplesFFT - 1]; // will do a this a bit later
//micDataSm = ((micData * 3) + micData)/4;
const int halfSamplesFFT = samplesFFT / 2; // samplesFFT divided by 2 const int halfSamplesFFT = samplesFFT / 2; // samplesFFT divided by 2
float maxSample1 = 0.0f; // max sample from first half of FFT batch float maxSample1 = 0.0f; // max sample from first half of FFT batch
float maxSample2 = 0.0f; // max sample from second half of FFT batch float maxSample2 = 0.0f; // max sample from second half of FFT batch
@ -219,34 +215,6 @@ void FFTcode(void * parameter)
// vReal[3 .. 255] contain useful data, each a 20Hz interval (60Hz - 5120Hz). // vReal[3 .. 255] contain useful data, each a 20Hz interval (60Hz - 5120Hz).
// There could be interesting data at bins 0 to 2, but there are too many artifacts. // There could be interesting data at bins 0 to 2, but there are too many artifacts.
// //
#ifdef MAJORPEAK_SUPPRESS_NOISE
// teporarily reduce signal strength in the highest + lowest bins
xtemp[0] = vReal[0]; vReal[0] *= 0.005f;
xtemp[1] = vReal[1]; vReal[1] *= 0.005f;
xtemp[2] = vReal[2]; vReal[2] *= 0.005f;
xtemp[3] = vReal[3]; vReal[3] *= 0.02f;
xtemp[4] = vReal[4]; vReal[4] *= 0.02f;
xtemp[5] = vReal[5]; vReal[5] *= 0.02f;
xtemp[6] = vReal[6]; vReal[6] *= 0.05f;
xtemp[7] = vReal[7]; vReal[7] *= 0.08f;
xtemp[8] = vReal[8]; vReal[8] *= 0.1f;
xtemp[9] = vReal[9]; vReal[9] *= 0.2f;
xtemp[10] = vReal[10]; vReal[10] *= 0.2f;
xtemp[11] = vReal[11]; vReal[11] *= 0.25f;
xtemp[12] = vReal[12]; vReal[12] *= 0.3f;
xtemp[13] = vReal[13]; vReal[13] *= 0.3f;
xtemp[14] = vReal[14]; vReal[14] *= 0.4f;
xtemp[15] = vReal[15]; vReal[15] *= 0.4f;
xtemp[16] = vReal[16]; vReal[16] *= 0.4f;
xtemp[17] = vReal[17]; vReal[17] *= 0.5f;
xtemp[18] = vReal[18]; vReal[18] *= 0.5f;
xtemp[19] = vReal[19]; vReal[19] *= 0.6f;
xtemp[20] = vReal[20]; vReal[20] *= 0.7f;
xtemp[21] = vReal[21]; vReal[21] *= 0.8f;
xtemp[22] = vReal[samplesFFT-2]; vReal[samplesFFT-2] = 0.0f;
xtemp[23] = vReal[samplesFFT-1]; vReal[samplesFFT-1] = 0.0f;
#endif
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT #ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant
@ -254,36 +222,6 @@ void FFTcode(void * parameter)
FFT.MajorPeak(&FFT_MajorPeak, &FFT_Magnitude); // let the effects know which freq was most dominant FFT.MajorPeak(&FFT_MajorPeak, &FFT_Magnitude); // let the effects know which freq was most dominant
#endif #endif
#ifdef MAJORPEAK_SUPPRESS_NOISE
// dirty hack: limit suppressed channel intensities to FFT_Magnitude
for (int k=0; k < 24; k++) if(xtemp[k] > FFT_Magnitude) xtemp[k] = FFT_Magnitude;
// restore bins
vReal[0] = xtemp[0];
vReal[1] = xtemp[1];
vReal[2] = xtemp[2];
vReal[3] = xtemp[3];
vReal[4] = xtemp[4];
vReal[5] = xtemp[5];
vReal[6] = xtemp[6];
vReal[7] = xtemp[7];
vReal[8] = xtemp[8];
vReal[9] = xtemp[9];
vReal[10] = xtemp[10];
vReal[11] = xtemp[11];
vReal[12] = xtemp[12];
vReal[13] = xtemp[13];
vReal[14] = xtemp[14];
vReal[15] = xtemp[15];
vReal[16] = xtemp[16];
vReal[17] = xtemp[17];
vReal[18] = xtemp[18];
vReal[19] = xtemp[19];
vReal[20] = xtemp[20];
vReal[21] = xtemp[21];
vReal[samplesFFT-2] = xtemp[22];
vReal[samplesFFT-1] = xtemp[23];
#endif
for (int i = 0; i < samplesFFT; i++) { // Values for bins 0 and 1 are WAY too large. Might as well start at 3. for (int i = 0; i < samplesFFT; i++) { // Values for bins 0 and 1 are WAY too large. Might as well start at 3.
float t = fabs(vReal[i]); // just to be sure - values in fft bins should be positive any way float t = fabs(vReal[i]); // just to be sure - values in fft bins should be positive any way
fftBin[i] = t / 16.0f; // Reduce magnitude. Want end result to be linear and ~4096 max. fftBin[i] = t / 16.0f; // Reduce magnitude. Want end result to be linear and ~4096 max.
@ -344,14 +282,6 @@ void FFTcode(void * parameter)
} }
#endif #endif
#ifdef SR_DEBUG
// Looking for fftResultMax for each bin using Pink Noise
for (int i=0; i<16; i++) {
fftResultMax[i] = ((fftResultMax[i] * 63.0f) + fftResult[i]) / 64.0f;
DEBUGSR_PRINT(fftResultMax[i]*fftResultPink[i]); DEBUGSR_PRINT("\t");
}
DEBUGSR_PRINTLN();
#endif
} // for(;;) } // for(;;)
} // FFTcode() } // FFTcode()
@ -482,32 +412,22 @@ class AudioReactive : public Usermod {
void logAudio() void logAudio()
{ {
#ifdef MIC_LOGGER #ifdef MIC_LOGGER
//Serial.print("micData:"); Serial.print(micData); Serial.print("\t"); // Debugging functions for audio input and sound processing. Comment out the values you want to see
//Serial.print("micDataSm:"); Serial.print(micDataSm); Serial.print("\t"); Serial.print("micReal:"); Serial.print(micDataReal); Serial.print("\t");
//Serial.print("micIn:"); Serial.print(micIn); Serial.print("\t"); //Serial.print("micData:"); Serial.print(micData); Serial.print("\t");
//Serial.print("micLev:"); Serial.print(micLev); Serial.print("\t"); //Serial.print("micDataSm:"); Serial.print(micDataSm); Serial.print("\t");
//Serial.print("sample:"); Serial.print(sample); Serial.print("\t"); //Serial.print("micIn:"); Serial.print(micIn); Serial.print("\t");
//Serial.print("sampleAvg:"); Serial.print(sampleAvg); Serial.print("\t"); //Serial.print("micLev:"); Serial.print(micLev); Serial.print("\t");
Serial.print("sampleReal:"); Serial.print(sampleReal); Serial.print("\t"); //Serial.print("sampleReal:"); Serial.print(sampleReal); Serial.print("\t");
//Serial.print("sampleMax:"); Serial.print(sampleMax); Serial.print("\t"); //Serial.print("sample:"); Serial.print(sample); Serial.print("\t");
Serial.print("multAgc:"); Serial.print(multAgc, 4); Serial.print("\t"); //Serial.print("sampleAvg:"); Serial.print(sampleAvg); Serial.print("\t");
Serial.print("sampleAgc:"); Serial.print(sampleAgc); Serial.print("\t"); //Serial.print("sampleMax:"); Serial.print(sampleMax); Serial.print("\t");
Serial.println(); //Serial.print("samplePeak:"); Serial.print((samplePeak!=0) ? 128:0); Serial.print("\t");
#endif //Serial.print("multAgc:"); Serial.print(multAgc, 4); Serial.print("\t");
Serial.print("sampleAgc:"); Serial.print(sampleAgc); Serial.print("\t");
//Serial.print("volumeRaw:"); Serial.print(volumeRaw); Serial.print("\t");
//Serial.print("volumeSmth:"); Serial.print(volumeSmth); Serial.print("\t");
#ifdef MIC_SAMPLING_LOG
//------------ Oscilloscope output ---------------------------
Serial.print(targetAgc); Serial.print(" ");
Serial.print(multAgc); Serial.print(" ");
Serial.print(sampleAgc); Serial.print(" ");
Serial.print(sample); Serial.print(" ");
Serial.print(sampleAvg); Serial.print(" ");
Serial.print(micLev); Serial.print(" ");
Serial.print(samplePeak); Serial.print(" "); //samplePeak = 0;
Serial.print(micIn); Serial.print(" ");
Serial.print(100); Serial.print(" ");
Serial.print(0); Serial.print(" ");
Serial.println(); Serial.println();
#endif #endif
@ -672,18 +592,12 @@ class AudioReactive : public Usermod {
micDataReal = micIn; micDataReal = micIn;
#else #else
micIn = micDataSm; // micDataSm = ((micData * 3) + micData)/4; micIn = micDataSm; // micDataSm = ((micData * 3) + micData)/4;
//DEBUGSR_PRINT("micIn:\tmicData:\tmicIn>>2:\tmic_In_abs:\tsample:\tsampleAdj:\tsampleAvg:\n");
//DEBUGSR_PRINT(micIn); DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(micData);
#endif #endif
// Note to self: the next line kills 80% of sample - "miclev" filter runs at "full arduino loop" speed, following the signal almost instantly!
//micLev = ((micLev * 31) + micIn) / 32; // Smooth it out over the last 32 samples for automatic centering
micLev = ((micLev * 8191.0f) + micDataReal) / 8192.0f; // takes a few seconds to "catch up" with the Mic Input micLev = ((micLev * 8191.0f) + micDataReal) / 8192.0f; // takes a few seconds to "catch up" with the Mic Input
if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal
micIn -= micLev; // Let's center it to 0 now micIn -= micLev; // Let's center it to 0 now
DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(micIn);
// Using an exponential filter to smooth out the signal. We'll add controls for this in a future release. // Using an exponential filter to smooth out the signal. We'll add controls for this in a future release.
float micInNoDC = fabs(micDataReal - micLev); float micInNoDC = fabs(micDataReal - micLev);
expAdjF = (weighting * micInNoDC + (1.0-weighting) * expAdjF); expAdjF = (weighting * micInNoDC + (1.0-weighting) * expAdjF);
@ -691,13 +605,9 @@ class AudioReactive : public Usermod {
expAdjF = fabsf(expAdjF); // Now (!) take the absolute value expAdjF = fabsf(expAdjF); // Now (!) take the absolute value
tmpSample = expAdjF; tmpSample = expAdjF;
DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(tmpSample);
micIn = abs(micIn); // And get the absolute value of each sample micIn = abs(micIn); // And get the absolute value of each sample
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 = sampleAdj;
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), 0); // Question: why are we limiting the value to 8 bits ???
@ -716,9 +626,6 @@ class AudioReactive : public Usermod {
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.
DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(sampleRaw);
DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(sampleAvg); DEBUGSR_PRINT("\n\n");
// Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC // Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC
uint16_t MinShowDelay = strip.getMinShowDelay(); uint16_t MinShowDelay = strip.getMinShowDelay();
@ -729,14 +636,6 @@ class AudioReactive : public Usermod {
//if (userVar1 == 0) samplePeak = 0; //if (userVar1 == 0) samplePeak = 0;
// Poor man's beat detection by seeing if sample > Average + some value. // Poor man's beat detection by seeing if sample > Average + some value.
// Serial.print(binNum); Serial.print("\t");
// Serial.print(fftBin[binNum]);
// Serial.print("\t");
// Serial.print(fftAvg[binNum/16]);
// Serial.print("\t");
// Serial.print(maxVol);
// Serial.print("\t");
// Serial.println(samplePeak);
if ((fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) { // This goes through ALL of the 255 bins if ((fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) { // This goes through ALL of the 255 bins
// if (sample > (sampleAvg + maxVol) && millis() > (timeOfPeak + 200)) { // if (sample > (sampleAvg + maxVol) && millis() > (timeOfPeak + 200)) {
// Then we got a peak, else we don't. The peak has to time out on its own in order to support UDP sound sync. // Then we got a peak, else we don't. The peak has to time out on its own in order to support UDP sound sync.
@ -960,7 +859,6 @@ class AudioReactive : public Usermod {
return; return;
} }
// We cannot wait indefinitely before processing audio data // We cannot wait indefinitely before processing audio data
//if (!enabled || strip.isUpdating()) return;
if (strip.isUpdating() && (millis() - lastUMRun < 2)) return; // be nice, but not too nice if (strip.isUpdating() && (millis() - lastUMRun < 2)) return; // be nice, but not too nice
// suspend local sound processing when "real time mode" is active (E131, UDP, ADALIGHT, ARTNET) // suspend local sound processing when "real time mode" is active (E131, UDP, ADALIGHT, ARTNET)