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optimization, and moving peak detection into own function

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- save 1K of RAM by optimizing out
 fftBin[].
- moved several copies of the peak reset code into a single function
- moved peak detection out of getSample().
 - call peak detection function as last step of FFTcode. More optimal, and we can be sure that fresh FFT result are available.

Peak detection/reset are now called from both tasks, so I had to move some peak-related vars out of AudioReactive class and make them global (static).
This commit is contained in:
Frank 2022-08-28 16:26:34 +02:00
parent 6019b7bda4
commit 64970772c7
1 changed files with 51 additions and 40 deletions
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91
usermods/audioreactive/audio_reactive.h
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@ -90,6 +90,15 @@ static float micDataReal = 0.0f; // MicIn data with full 24bit re
static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier
static float sampleAvg = 0.0f; // Smoothed Average sample - sampleAvg < 1 means "quiet" (simple noise gate)
// peak detection
static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay()
static uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
static uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated)
static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
static unsigned long timeOfPeak = 0; // time of last sample peak detection.
static void detectSamplePeak(void); // peak detection function (needs scaled FFT reasults in vReal[])
static void autoResetPeak(void); // peak auto-reset function
////////////////////
// Begin FFT Code //
@ -105,7 +114,7 @@ static float sampleAvg = 0.0f; // Smoothed Average sample - sam
// FFT Output variables shared with animations
#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !!
static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency
static float FFT_Magnitude = 0.0f; // FFT: magintude peak frequency
static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency
static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects
// FFT Constants
@ -115,7 +124,6 @@ constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT resul
// These are the input and output vectors. Input vectors receive computed results from FFT.
static float vReal[samplesFFT] = {0.0f}; // FFT sample inputs / freq output - these are our raw result bins
static float vImag[samplesFFT] = {0.0f}; // imaginary parts
static float fftBin[samplesFFT_2] = {0.0f};
// the following are observed values, supported by a bit of "educated guessing"
//#define FFT_DOWNSCALE 0.65f // 20kHz - downscaling factor for FFT results - "Flat-Top" window @20Khz, old freq channels
@ -159,7 +167,7 @@ static float mapf(float x, float in_min, float in_max, float out_min, float out_
static float fftAddAvg(int from, int to) {
float result = 0.0f;
for (int i = from; i <= to; i++) {
result += fftBin[i];
result += vReal[i];
}
return result / float(to - from + 1);
}
@ -235,9 +243,9 @@ void FFTcode(void * parameter)
#endif
FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
for (int i = 0; i < samplesFFT_2; i++) { // Values for bins 0 and 1 are WAY too large. Might as well start at 3.
for (int i = 0; i < samplesFFT; i++) {
float t = fabsf(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.
vReal[i] = t / 16.0f; // Reduce magnitude. Want end result to be scaled linear and ~4096 max.
} // for()
// mapping of FFT result bins to frequency channels
@ -375,11 +383,40 @@ void FFTcode(void * parameter)
unsigned long fftTimeInMillis = ((esp_timer_get_time() - start) +500ULL) / 1000ULL; // "+500" to ensure proper rounding
fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth
}
#endif
#endif
// run peak detection
autoResetPeak();
detectSamplePeak();
} // for(;;)ever
} // FFTcode() task end
////////////////////
// Peak detection //
////////////////////
// peak detection is called from FFT task when vReal[] contains valid FFT results
static void detectSamplePeak(void) {
// Poor man's beat detection by seeing if sample > Average + some value.
if ((sampleAvg > 1) && (maxVol > 0) && (binNum > 1) && (vReal[binNum] > maxVol) && ((millis() - timeOfPeak) > 100)) {
// This goes through ALL of the 255 bins - but ignores stupid settings
// 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.
samplePeak = true;
timeOfPeak = millis();
udpSamplePeak = true;
}
}
static void autoResetPeak(void) {
uint16_t MinShowDelay = MAX(50, strip.getMinShowDelay()); // Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC
if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
samplePeak = false;
if (audioSyncEnabled == 0) udpSamplePeak = false; // this is normally reset by transmitAudioData
}
}
//class name. Use something descriptive and leave the ": public Usermod" part :)
class AudioReactive : public Usermod {
@ -479,12 +516,6 @@ class AudioReactive : public Usermod {
float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample
int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc
float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc
// peak detection
uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated)
bool samplePeak = false; // Boolean flag for peak. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay()
bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
unsigned long timeOfPeak = 0; // time of last sample peak detection
// used to feed "Info" Page
unsigned long last_UDPTime = 0; // time of last valid UDP sound sync datapacket
@ -728,24 +759,6 @@ class AudioReactive : public Usermod {
if (sampleMax < 0.5f) sampleMax = 0.0f;
sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
// Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC
uint16_t MinShowDelay = strip.getMinShowDelay();
if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
samplePeak = false;
udpSamplePeak = false;
}
//if (userVar1 == 0) samplePeak = 0;
// Poor man's beat detection by seeing if sample > Average + some value.
if ((maxVol > 0) && (binNum > 1) && (fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) {
// This goes through ALL of the 255 bins - but ignores stupid settings
// 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.
samplePeak = true;
timeOfPeak = millis();
udpSamplePeak = true;
}
} // getSample()
@ -838,13 +851,7 @@ class AudioReactive : public Usermod {
sampleAgc = volumeSmth;
multAgc = 1.0f;
// auto-reset sample peak. Need to do it here, because getSample() is not running
uint16_t MinShowDelay = strip.getMinShowDelay();
if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
samplePeak = false;
udpSamplePeak = false;
}
//if (userVar1 == 0) samplePeak = 0;
autoResetPeak();
// Only change samplePeak IF it's currently false.
// If it's true already, then the animation still needs to respond.
if (!samplePeak) {
@ -1066,9 +1073,11 @@ class AudioReactive : public Usermod {
if (soundAgc) my_magnitude *= multAgc;
if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
limitSampleDynamics(); // optional - makes volumeSmth very smooth and fluent
}
limitSampleDynamics();
} // if (!disableSoundProcessing)
autoResetPeak(); // auto-reset sample peak after strip minShowDelay
if (!udpSyncConnected) udpSamplePeak = false; // reset UDP samplePeak while UDP is unconnected
// UDP Microphone Sync - receive mode
if ((audioSyncEnabled & 0x02) && udpSyncConnected) {
@ -1091,7 +1100,7 @@ class AudioReactive : public Usermod {
}
#endif
// peak sample from last 5 seconds
// Info Page: keep max sample from last 5 seconds
if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) {
sampleMaxTimer = millis();
maxSample5sec = (0.15 * maxSample5sec) + 0.85 *((soundAgc) ? sampleAgc : sampleAvg); // reset, and start with some smoothing
@ -1099,6 +1108,7 @@ class AudioReactive : public Usermod {
} else {
if ((sampleAvg >= 1)) maxSample5sec = fmaxf(maxSample5sec, (soundAgc) ? rawSampleAgc : sampleRaw); // follow maximum volume
}
//UDP Microphone Sync - transmit mode
if ((audioSyncEnabled & 0x01) && (millis() - lastTime > 20)) {
// Only run the transmit code IF we're in Transmit mode
@ -1138,6 +1148,7 @@ class AudioReactive : public Usermod {
memset(fftResult, 0, sizeof(fftResult));
for(int i=(init?0:1); i<NUM_GEQ_CHANNELS; i+=2) fftResult[i] = 16; // make a tiny pattern
inputLevel = 128; // resset level slider to default
autoResetPeak();
if (init && FFT_Task) {
vTaskSuspend(FFT_Task); // update is about to begin, disable task to prevent crash