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some audio processing improvements and bugfixes from SR WLED

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- smoothing FFTResult (don't have a matrix to test)
- UDP sound sync improvements
- some bugfixes from SR WLED
- button.cpp: avoid starvation: strip.isUpdating() can be true for a long time.

work in progress - still needs testing!!
This commit is contained in:
Frank
2022-08-14 13:58:07 +02:00
parent d05b49496c
commit 968721a515
4 changed files with 62 additions and 30 deletions
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72
usermods/audioreactive/audio_reactive.h
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@@ -23,10 +23,13 @@
// Comment/Uncomment to toggle usb serial debugging
// #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter)
// #define FFT_SAMPLING_LOG // FFT result debugging
// #define SR_DEBUG // generic SR DEBUG messages
// #define SR_DEBUG // generic SR DEBUG messages (including MIC_LOGGER)
// #define NO_MIC_LOGGER // exclude MIC_LOGGER from SR_DEBUG
// hackers corner
//#define SOUND_DYNAMICS_LIMITER // experimental: define to enable a dynamics limiter that avoids "sudden flashes" at onsets. Makes some effects look more "smooth and fluent"
#if !defined(SOUND_DYNAMICS_LIMITER) && !defined(NO_SOUND_DYNAMICS_LIMITER)
#define SOUND_DYNAMICS_LIMITER // experimental: define to enable a dynamics limiter that avoids "sudden flashes" at onsets. Makes some effects look more "smooth and fluent"
#endif
#ifdef SR_DEBUG
#define DEBUGSR_PRINT(x) Serial.print(x)
@@ -60,6 +63,10 @@ static uint8_t sampleGain = 60; // sample gain (config value)
static uint8_t soundAgc = 0; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value)
static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1 - receive (config value)
// user settable parameters for limitSoundDynamics()
static int attackTime = 80; // int: attack time in milliseconds. Default 0.1sec
static int decayTime = 1400; // int: decay time in milliseconds. Default 1.4sec
//
// AGC presets
// Note: in C++, "const" implies "static" - no need to explicitly declare everything as "static const"
@@ -98,7 +105,7 @@ static float multAgc = 1.0f; // sample * multAgc = sampleAgc.
// FFT Variables
constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - This value MUST ALWAYS be a power of 2
constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - nly the "lower half" contains usefull information.
constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information.
static float FFT_MajorPeak = 0.0f;
static float FFT_Magnitude = 0.0f;
@@ -274,9 +281,12 @@ void FFTcode(void * parameter)
// Manual linear adjustment of gain using sampleGain adjustment for different input types.
fftCalc[i] *= soundAgc ? multAgc : ((float)sampleGain/40.0f * (float)inputLevel/128.0f + 1.0f/16.0f); //with inputLevel adjustment
// smooth results
//fftAvg[i] = fftCalc[i]*0.05f + 0.95f*fftAvg[i]; // will need approx 10 cycles (250ms) for converging against fftCalc[i]
fftAvg[i] = fftCalc[i] *0.1f + 0.9f*fftAvg[i]; // will need approx 5 cycles (125ms) for converging against fftCalc[i]
// Now, let's dump it all into fftResult. Need to do this, otherwise other routines might grab fftResult values prematurely.
fftResult[i] = constrain((int)fftCalc[i], 0, 254);
fftAvg[i] = (float)fftResult[i]*0.05f + 0.95f*fftAvg[i];
//fftResult[i] = constrain((int)fftCalc[i], 0, 254);
fftResult[i] = constrain((int)fftAvg[i], 0, 254);
}
#ifdef WLED_DEBUG
@@ -602,10 +612,13 @@ class AudioReactive : public Usermod {
// 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.
static unsigned long lastAnalogTime = 0;
static float lastAnalogValue = 0.0f;
if (millis() - lastAnalogTime > 20) {
micDataReal = analogRead(A0); // read one sample with 10bit resolution. This is a dirty hack, supporting volumereactive effects only.
lastAnalogTime = millis();
}
lastAnalogValue = micDataReal;
yield();
} else micDataReal = lastAnalogValue;
micIn = int(micDataReal);
#endif
#endif
@@ -618,6 +631,7 @@ class AudioReactive : public Usermod {
float micInNoDC = fabs(micDataReal - micLev);
expAdjF = (weighting * micInNoDC + (1.0-weighting) * expAdjF);
expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate
if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0"
expAdjF = fabsf(expAdjF); // Now (!) take the absolute value
tmpSample = expAdjF;
@@ -664,14 +678,9 @@ class AudioReactive : public Usermod {
/* Limits the dynamics of volumeSmth (= sampleAvg or sampleAgc).
* It does not affect FFTResult[] or volumeRaw ( = sample or rawSampleAgc)
* does not affect FFTResult[] or volumeRaw ( = sample or rawSampleAgc)
*/
// effects: Gravimeter, Gravcenter, Gravcentric, Noisefire, Plasmoid, Freqpixels, Freqwave, Gravfreq, (2D Swirl, 2D Waverly)
// experimental, as it still has side-effects on AGC - AGC detects "silence" to late (due to long decay time) and ditches up the gain multiplier.
// values below will be made user-configurable later
const float attackTime = 200; // attack time -> 0.2sec
const float decayTime = 2800; // decay time -> 2.8sec
void limitSampleDynamics(void) {
#ifdef SOUND_DYNAMICS_LIMITER
const float bigChange = 196; // just a representative number - a large, expected sample value
@@ -681,8 +690,8 @@ class AudioReactive : public Usermod {
if ((attackTime > 0) && (decayTime > 0)) { // only change volume if user has defined attack>0 and decay>0
long delta_time = millis() - last_time;
delta_time = constrain(delta_time , 1, 1000); // below 1ms -> 1ms; above 1sec -> sily lil hick-up
float maxAttack = bigChange * float(delta_time) / attackTime;
float maxDecay = - bigChange * float(delta_time) / decayTime;
float maxAttack = bigChange * float(delta_time) / float(attackTime);
float maxDecay = - bigChange * float(delta_time) / float(decayTime);
float deltaSample = volumeSmth - last_volumeSmth;
if (deltaSample > maxAttack) deltaSample = maxAttack;
@@ -704,8 +713,11 @@ class AudioReactive : public Usermod {
audioSyncPacket transmitData;
strncpy_P(transmitData.header, PSTR(UDP_SYNC_HEADER), 6);
transmitData.sampleRaw = volumeRaw;
transmitData.sampleSmth = volumeSmth;
//transmitData.sampleRaw = volumeRaw;
//transmitData.sampleSmth = volumeSmth;
// transmit samples that were not modified by limitSampleDynamics()
transmitData.sampleRaw = (soundAgc) ? rawSampleAgc: sampleRaw;
transmitData.sampleSmth = (soundAgc) ? sampleAgc : sampleAvg;
transmitData.samplePeak = udpSamplePeak ? 1:0;
udpSamplePeak = false; // Reset udpSamplePeak after we've transmitted it
transmitData.reserved1 = 0;
@@ -744,9 +756,11 @@ class AudioReactive : public Usermod {
if (packetSize == sizeof(audioSyncPacket) && !(isValidUdpSyncVersion((const char *)fftBuff))) {
audioSyncPacket *receivedPacket = reinterpret_cast<audioSyncPacket*>(fftBuff);
// update samples for effects
volumeSmth = receivedPacket->sampleSmth;
volumeRaw = receivedPacket->sampleRaw;
// update internal samples
sampleRaw = volumeRaw;
sampleAvg = volumeSmth;
rawSampleAgc = volumeRaw;
@@ -945,6 +959,7 @@ class AudioReactive : public Usermod {
if (audioSyncEnabled & 0x02) disableSoundProcessing = true; // make sure everything is disabled IF in audio Receive mode
if (audioSyncEnabled & 0x01) disableSoundProcessing = false; // keep running audio IF we're in audio Transmit mode
// Only run the sampling code IF we're not in Receive mode or realtime mode
if (!(audioSyncEnabled & 0x02) && !disableSoundProcessing) {
bool agcEffect = false;
@@ -981,9 +996,8 @@ class AudioReactive : public Usermod {
limitSampleDynamics(); // optional - makes volumeSmth very smooth and fluent
// update UI
// update WebServer UI
uint8_t knownMode = strip.getFirstSelectedSeg().mode; // 1st selected segment is more appropriate than main segment
if (lastMode != knownMode) { // only execute if mode changes
char lineBuffer[4];
extractModeName(knownMode, JSON_mode_names, lineBuffer, 3); // use of JSON_mode_names is deprecated, use nullptr
@@ -1024,10 +1038,19 @@ class AudioReactive : public Usermod {
}
}
// Begin UDP Microphone Sync
if ((audioSyncEnabled & 0x02) && millis() - lastTime > delayMs) { // Only run the audio listener code if we're in Receive mode
(void) receiveAudioData(); // ToDo: use return value for something meaningfull
lastTime = millis();
// UDP Microphone Sync - receive mode
if ((audioSyncEnabled & 0x02) && udpSyncConnected) {
// Only run the audio listener code if we're in Receive mode
static float syncVolumeSmth = 0;
bool have_new_sample = false;
if (millis() - lastTime > delayMs) {
have_new_sample = receiveAudioData();
lastTime = millis();
}
if (have_new_sample) syncVolumeSmth = volumeSmth; // remember received sample
else volumeSmth = syncVolumeSmth; // restore originally received sample for next run of dynamics limiter
limitSampleDynamics(); // run dynamics limiter on received volumeSmth, to hide jumps and hickups
}
#if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
@@ -1036,12 +1059,13 @@ class AudioReactive : public Usermod {
}
#endif
if ((audioSyncEnabled & 0x01) && millis() - lastTime > 20) { // Only run the transmit code IF we're in Transmit mode
//UDP Microphone Sync - transmit mode
if ((audioSyncEnabled & 0x01) && millis() - lastTime > 20) {
// Only run the transmit code IF we're in Transmit mode
transmitAudioData();
lastTime = millis();
}
//limitSampleDynamics(); // If done as the last step, it will also affect audio received by UDP sound sync. Problem: effects might see inconsistent intermediate values and start flickering :-(
}