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minor updates for audioreactive

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* remove "not supported" warning on -S2
* minor tweaking of beat detector
* optimized parameters for arduinoFFT
* fixed a few implicit "double" promotions
* minor UDP protocol optimizations
* small optimization for fft task create / suspend
* completely remove analogmic settings for -S3/-S2/-C3 where analog is not supported

changes were already tested extensively in the MM fork.
This commit is contained in:
Frank
2023-09-09 21:07:20 +02:00
parent da0cf01eee
commit e9723de6c3
2 changed files with 57 additions and 38 deletions
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84
usermods/audioreactive/audio_reactive.h
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@@ -20,6 +20,12 @@
* ....
*/
#if !defined(FFTTASK_PRIORITY)
#define FFTTASK_PRIORITY 1 // standard: looptask prio
//#define FFTTASK_PRIORITY 2 // above looptask, below asyc_tcp
//#define FFTTASK_PRIORITY 4 // above asyc_tcp
#endif
// Comment/Uncomment to toggle usb serial debugging
// #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter)
// #define FFT_SAMPLING_LOG // FFT result debugging
@@ -104,7 +110,7 @@ static float sampleAgc = 0.0f; // Smoothed AGC sample
// 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 maxVol = 31; // 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.
@@ -173,13 +179,18 @@ static float windowWeighingFactors[samplesFFT] = {0.0f};
// Create FFT object
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
// lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2
#define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc), and an a few other speedups
#define FFT_SQRT_APPROXIMATION // enables "quake3" style inverse sqrt
#define sqrt(x) sqrtf(x) // little hack that reduces FFT time by 50% on ESP32 (as alternative to FFT_SQRT_APPROXIMATION)
// lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2
// these options actually cause slow-downs on all esp32 processors, don't use them.
// #define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc) - not faster on ESP32
// #define FFT_SQRT_APPROXIMATION // enables "quake3" style inverse sqrt - slower on ESP32
// Below options are forcing ArduinoFFT to use sqrtf() instead of sqrt()
#define sqrt(x) sqrtf(x) // little hack that reduces FFT time by 10-50% on ESP32
#define sqrt_internal sqrtf // see https://github.com/kosme/arduinoFFT/pull/83
#else
// lib_deps += https://github.com/blazoncek/arduinoFFT.git
// around 40% slower on -S2
// lib_deps += https://github.com/blazoncek/arduinoFFT.git
#endif
#include <arduinoFFT.h>
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
@@ -411,7 +422,7 @@ static void runMicFilter(uint16_t numSamples, float *sampleBuffer) // p
//constexpr float beta1 = 0.8285f; // 18Khz
constexpr float beta1 = 0.85f; // 20Khz
constexpr float beta2 = (1.0f - beta1) / 2.0;
constexpr float beta2 = (1.0f - beta1) / 2.0f;
static float last_vals[2] = { 0.0f }; // FIR high freq cutoff filter
static float lowfilt = 0.0f; // IIR low frequency cutoff filter
@@ -464,17 +475,17 @@ static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels) // p
switch (FFTScalingMode) {
case 1:
// Logarithmic scaling
currentResult *= 0.42; // 42 is the answer ;-)
currentResult -= 8.0; // this skips the lowest row, giving some room for peaks
if (currentResult > 1.0) currentResult = logf(currentResult); // log to base "e", which is the fastest log() function
else currentResult = 0.0; // special handling, because log(1) = 0; log(0) = undefined
currentResult *= 0.42f; // 42 is the answer ;-)
currentResult -= 8.0f; // this skips the lowest row, giving some room for peaks
if (currentResult > 1.0f) currentResult = logf(currentResult); // log to base "e", which is the fastest log() function
else currentResult = 0.0f; // special handling, because log(1) = 0; log(0) = undefined
currentResult *= 0.85f + (float(i)/18.0f); // extra up-scaling for high frequencies
currentResult = mapf(currentResult, 0, LOG_256, 0, 255); // map [log(1) ... log(255)] to [0 ... 255]
break;
case 2:
// Linear scaling
currentResult *= 0.30f; // needs a bit more damping, get stay below 255
currentResult -= 4.0; // giving a bit more room for peaks
currentResult -= 4.0f; // giving a bit more room for peaks
if (currentResult < 1.0f) currentResult = 0.0f;
currentResult *= 0.85f + (float(i)/1.8f); // extra up-scaling for high frequencies
break;
@@ -482,8 +493,8 @@ static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels) // p
// square root scaling
currentResult *= 0.38f;
currentResult -= 6.0f;
if (currentResult > 1.0) currentResult = sqrtf(currentResult);
else currentResult = 0.0; // special handling, because sqrt(0) = undefined
if (currentResult > 1.0f) currentResult = sqrtf(currentResult);
else currentResult = 0.0f; // special handling, because sqrt(0) = undefined
currentResult *= 0.85f + (float(i)/4.5f); // extra up-scaling for high frequencies
currentResult = mapf(currentResult, 0.0, 16.0, 0.0, 255.0); // map [sqrt(1) ... sqrt(256)] to [0 ... 255]
break;
@@ -511,11 +522,11 @@ static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels) // p
// peak detection is called from FFT task when vReal[] contains valid FFT results
static void detectSamplePeak(void) {
bool havePeak = false;
// softhack007: this code continuously triggers while amplitude in the selected bin is above a certain threshold. So it does not detect peaks - it detects high activity in a frequency bin.
// Poor man's beat detection by seeing if sample > Average + some value.
// 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.
if ((sampleAvg > 1) && (maxVol > 0) && (binNum > 1) && (vReal[binNum] > maxVol) && ((millis() - timeOfPeak) > 100)) {
if ((sampleAvg > 1) && (maxVol > 0) && (binNum > 4) && (vReal[binNum] > maxVol) && ((millis() - timeOfPeak) > 100)) {
havePeak = true;
}
@@ -758,7 +769,7 @@ class AudioReactive : public Usermod {
if (time_now - last_time > 2) {
last_time = time_now;
if((fabsf(sampleReal) < 2.0f) || (sampleMax < 1.0f)) {
if((fabsf(sampleReal) < 2.0f) || (sampleMax < 1.0)) {
// MIC signal is "squelched" - deliver silence
tmpAgc = 0;
// we need to "spin down" the intgrated error buffer
@@ -873,8 +884,8 @@ class AudioReactive : public Usermod {
// keep "peak" sample, but decay value if current sample is below peak
if ((sampleMax < sampleReal) && (sampleReal > 0.5f)) {
sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // new peak - with some filtering
// another simple way to detect samplePeak
if ((binNum < 10) && (millis() - timeOfPeak > 80) && (sampleAvg > 1)) {
// another simple way to detect samplePeak - cannot detect beats, but reacts on peak volume
if (((binNum < 12) || ((maxVol < 1))) && (millis() - timeOfPeak > 80) && (sampleAvg > 1)) {
samplePeak = true;
timeOfPeak = millis();
udpSamplePeak = true;
@@ -964,9 +975,10 @@ class AudioReactive : public Usermod {
transmitData.FFT_Magnitude = my_magnitude;
transmitData.FFT_MajorPeak = FFT_MajorPeak;
fftUdp.beginMulticastPacket();
fftUdp.write(reinterpret_cast<uint8_t *>(&transmitData), sizeof(transmitData));
fftUdp.endPacket();
if (fftUdp.beginMulticastPacket() != 0) { // beginMulticastPacket returns 0 in case of error
fftUdp.write(reinterpret_cast<uint8_t *>(&transmitData), sizeof(transmitData));
fftUdp.endPacket();
}
return;
} // transmitAudioData()
@@ -1272,9 +1284,10 @@ class AudioReactive : public Usermod {
#ifdef WLED_DEBUG
// complain when audio userloop has been delayed for long time. Currently we need userloop running between 500 and 1500 times per second.
if ((userloopDelay > 23) && !disableSoundProcessing && (audioSyncEnabled == 0)) {
DEBUG_PRINTF("[AR userLoop] hickup detected -> was inactive for last %d millis!\n", userloopDelay);
}
// softhack007 disabled temporarily - avoid serial console spam with MANY leds and low FPS
//if ((userloopDelay > 65) && !disableSoundProcessing && (audioSyncEnabled == 0)) {
//DEBUG_PRINTF("[AR userLoop] hickup detected -> was inactive for last %d millis!\n", userloopDelay);
//}
#endif
// run filters, and repeat in case of loop delays (hick-up compensation)
@@ -1311,6 +1324,9 @@ class AudioReactive : public Usermod {
if (millis() - lastTime > delayMs) {
have_new_sample = receiveAudioData();
if (have_new_sample) last_UDPTime = millis();
#ifdef ARDUINO_ARCH_ESP32
else fftUdp.flush(); // Flush udp input buffers if we haven't read it - avoids hickups in receive mode. Does not work on 8266.
#endif
lastTime = millis();
}
if (have_new_sample) syncVolumeSmth = volumeSmth; // remember received sample
@@ -1329,7 +1345,7 @@ class AudioReactive : public Usermod {
// 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
maxSample5sec = (0.15f * maxSample5sec) + 0.85f *((soundAgc) ? sampleAgc : sampleAvg); // reset, and start with some smoothing
if (sampleAvg < 1) maxSample5sec = 0; // noise gate
} else {
if ((sampleAvg >= 1)) maxSample5sec = fmaxf(maxSample5sec, (soundAgc) ? rawSampleAgc : sampleRaw); // follow maximum volume
@@ -1373,10 +1389,11 @@ class AudioReactive : public Usermod {
memset(fftAvg, 0, sizeof(fftAvg));
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
inputLevel = 128; // reset level slider to default
autoResetPeak();
if (init && FFT_Task) {
delay(25); // give some time for I2S driver to finish sampling before we suspend it
vTaskSuspend(FFT_Task); // update is about to begin, disable task to prevent crash
if (udpSyncConnected) { // close UDP sync connection (if open)
udpSyncConnected = false;
@@ -1388,15 +1405,14 @@ class AudioReactive : public Usermod {
vTaskResume(FFT_Task);
connected(); // resume UDP
} else
// xTaskCreatePinnedToCore(
xTaskCreate( // no need to "pin" this task to core #0
xTaskCreateUniversal( // xTaskCreateUniversal also works on -S2 and -C3 with single core
FFTcode, // Function to implement the task
"FFT", // Name of the task
5000, // Stack size in words
NULL, // Task input parameter
1, // Priority of the task
FFTTASK_PRIORITY, // Priority of the task
&FFT_Task // Task handle
// , 0 // Core where the task should run
, 0 // Core where the task should run
);
}
micDataReal = 0.0f; // just to be sure
@@ -1493,7 +1509,7 @@ class AudioReactive : public Usermod {
infoArr.add(F("I2S digital"));
}
// input level or "silence"
if (maxSample5sec > 1.0) {
if (maxSample5sec > 1.0f) {
float my_usage = 100.0f * (maxSample5sec / 255.0f);
snprintf_P(myStringBuffer, 15, PSTR(" - peak %3d%%"), int(my_usage));
infoArr.add(myStringBuffer);
@@ -1503,7 +1519,7 @@ class AudioReactive : public Usermod {
} else {
// error during audio source setup
infoArr.add(F("not initialized"));
infoArr.add(F(" - check GPIO config"));
infoArr.add(F(" - check pin settings"));
}
}
@@ -1807,7 +1823,9 @@ class AudioReactive : public Usermod {
const char AudioReactive::_name[] PROGMEM = "AudioReactive";
const char AudioReactive::_enabled[] PROGMEM = "enabled";
const char AudioReactive::_inputLvl[] PROGMEM = "inputLevel";
#if defined(ARDUINO_ARCH_ESP32) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
const char AudioReactive::_analogmic[] PROGMEM = "analogmic";
#endif
const char AudioReactive::_digitalmic[] PROGMEM = "digitalmic";
const char AudioReactive::UDP_SYNC_HEADER[] PROGMEM = "00002"; // new sync header version, as format no longer compatible with previous structure
const char AudioReactive::UDP_SYNC_HEADER_v1[] PROGMEM = "00001"; // old sync header version - need to add backwards-compatibility feature