lba.gitea/WLED
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

AR: removed two unneeded variables

Browse Source 
some cleanup - no functional impact.
This commit is contained in:
Frank 2022-08-08 10:51:46 +02:00
parent 58987989da
commit 3a8c99d43c
1 changed files with 8 additions and 19 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

27
usermods/audioreactive/audio_reactive.h
View File

@ -82,9 +82,7 @@ const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; //
static AudioSource *audioSource = nullptr; static AudioSource *audioSource = nullptr;
static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks. static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks.
//static uint16_t micData; // Analog input for FFT static float micDataReal = 0.0f; // MicIn data with full 24bit resolution - lowest 8bit after decimal point
static uint16_t micDataSm; // Smoothed mic data, as it's a bit twitchy
static float micDataReal = 0.0f; // future support - this one has the full 24bit MicIn data - lowest 8bit after decimal point
static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier
//////////////////// ////////////////////
@ -201,7 +199,6 @@ void FFTcode(void * parameter)
if (fabsf((float)vReal[i]) > maxSample2) maxSample2 = fabsf((float)vReal[i]); if (fabsf((float)vReal[i]) > maxSample2) maxSample2 = fabsf((float)vReal[i]);
} }
// release first sample to volume reactive effects // release first sample to volume reactive effects
micDataSm = (uint16_t)maxSample1;
micDataReal = maxSample1; micDataReal = maxSample1;
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT #ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
@ -288,7 +285,6 @@ void FFTcode(void * parameter)
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, by waiting until FFT_MIN_CYCLE is over vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, by waiting until FFT_MIN_CYCLE is over
// release second sample to volume reactive effects. // release second sample to volume reactive effects.
// Releasing a second sample now effectively doubles the "sample rate" // Releasing a second sample now effectively doubles the "sample rate"
micDataSm = (uint16_t)maxSample2;
micDataReal = maxSample2; micDataReal = maxSample2;
} // for(;;) } // for(;;)
@ -373,15 +369,13 @@ class AudioReactive : public Usermod {
const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED
// variables used in effects // variables used in effects
uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger 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. uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated)
bool samplePeak = 0; // Boolean flag for peak. Responding routine must reset this flag bool samplePeak = 0; // Boolean flag for peak. Responding routine must reset this flag
float volumeSmth; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample float volumeSmth; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample
int16_t volumeRaw; // either sampleRaw or rawSampleAgc depending on soundAgc int16_t volumeRaw; // either sampleRaw or rawSampleAgc depending on soundAgc
float my_magnitude; // FFT_Magnitude, scaled by multAgc
#ifdef MIC_SAMPLING_LOG
uint8_t targetAgc = 60; // This is our setPoint at 20% of max for the adjusted output (used only in logAudio())
#endif
bool udpSamplePeak = 0; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData bool udpSamplePeak = 0; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
int16_t micIn = 0; // Current sample starts with negative values and large values, which is why it's 16 bit signed int16_t micIn = 0; // Current sample starts with negative values and large values, which is why it's 16 bit signed
int16_t sampleRaw; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel; smoothed over 16 samples) int16_t sampleRaw; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel; smoothed over 16 samples)
@ -415,16 +409,12 @@ class AudioReactive : public Usermod {
static const char UDP_SYNC_HEADER[]; static const char UDP_SYNC_HEADER[];
static const char UDP_SYNC_HEADER_v1[]; static const char UDP_SYNC_HEADER_v1[];
float my_magnitude;
// private methods // private methods
void logAudio() void logAudio()
{ {
#ifdef MIC_LOGGER #ifdef MIC_LOGGER
// Debugging functions for audio input and sound processing. Comment out the values you want to see // Debugging functions for audio input and sound processing. Comment out the values you want to see
Serial.print("micReal:"); Serial.print(micDataReal); Serial.print("\t"); Serial.print("micReal:"); Serial.print(micDataReal); Serial.print("\t");
//Serial.print("micData:"); Serial.print(micData); Serial.print("\t");
//Serial.print("micDataSm:"); Serial.print(micDataSm); Serial.print("\t");
//Serial.print("micIn:"); Serial.print(micIn); Serial.print("\t"); //Serial.print("micIn:"); Serial.print(micIn); Serial.print("\t");
//Serial.print("micLev:"); Serial.print(micLev); 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");
@ -474,7 +464,7 @@ class AudioReactive : public Usermod {
} }
for(int i = 0; i < 16; i++) { for(int i = 0; i < 16; i++) {
Serial.print(i); Serial.print(":"); Serial.print(i); Serial.print(":");
Serial.printf("%04d ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1)); Serial.printf("%04ld ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1));
} }
if(printMaxVal) { if(printMaxVal) {
Serial.printf("maxVal:%04d ", maxVal + (mapValuesToPlotterSpace ? 16*256 : 0)); Serial.printf("maxVal:%04d ", maxVal + (mapValuesToPlotterSpace ? 16*256 : 0));
@ -603,7 +593,7 @@ class AudioReactive : public Usermod {
micDataReal = micIn; micDataReal = micIn;
#else #else
#ifdef ESP32 #ifdef ESP32
micIn = micDataSm; // micDataSm = ((micData * 3) + micData)/4; micIn = int(micDataReal); // micDataSm = ((micData * 3) + micData)/4;
#else #else
// this is the minimal code for reading analog mic input on 8266. // this is the minimal code for reading analog mic input on 8266.
// warning!! Absolutely experimental code. Audio on 8266 is still not working. Expects a million follow-on problems. // warning!! Absolutely experimental code. Audio on 8266 is still not working. Expects a million follow-on problems.
@ -612,8 +602,7 @@ class AudioReactive : public Usermod {
micDataReal = analogRead(A0); // read one sample with 10bit resolution. This is a dirty hack, supporting volumereactive effects only. micDataReal = analogRead(A0); // read one sample with 10bit resolution. This is a dirty hack, supporting volumereactive effects only.
lastAnalogTime = millis(); lastAnalogTime = millis();
} }
micDataSm = micDataReal; micIn = int(micDataReal);
micIn = micDataSm;
#endif #endif
#endif #endif
@ -659,8 +648,8 @@ 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.
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)) {
if ((fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) { // This goes through ALL of the 255 bins
// 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.
samplePeak = true; samplePeak = true;
timeOfPeak = millis(); timeOfPeak = millis();