kopia lustrzana https://github.com/Aircoookie/WLED
some audio processing improvements and bugfixes from SR WLED
- 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!!pull/2737/head
Frank
rodzic
d05b49496c
commit
968721a515
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@ -23,10 +23,13 @@
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// Comment/Uncomment to toggle usb serial debugging
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// #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter)
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// #define FFT_SAMPLING_LOG // FFT result debugging
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// #define SR_DEBUG // generic SR DEBUG messages
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// #define SR_DEBUG // generic SR DEBUG messages (including MIC_LOGGER)
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// #define NO_MIC_LOGGER // exclude MIC_LOGGER from SR_DEBUG
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// hackers corner
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//#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"
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#if !defined(SOUND_DYNAMICS_LIMITER) && !defined(NO_SOUND_DYNAMICS_LIMITER)
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#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"
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#endif
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#ifdef SR_DEBUG
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#define DEBUGSR_PRINT(x) Serial.print(x)
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@ -60,6 +63,10 @@ static uint8_t sampleGain = 60; // sample gain (config value)
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static uint8_t soundAgc = 0; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value)
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static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1 - receive (config value)
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// user settable parameters for limitSoundDynamics()
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static int attackTime = 80; // int: attack time in milliseconds. Default 0.1sec
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static int decayTime = 1400; // int: decay time in milliseconds. Default 1.4sec
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//
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// AGC presets
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// Note: in C++, "const" implies "static" - no need to explicitly declare everything as "static const"
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@ -98,7 +105,7 @@ static float multAgc = 1.0f; // sample * multAgc = sampleAgc.
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// FFT Variables
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constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - This value MUST ALWAYS be a power of 2
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constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - nly the "lower half" contains usefull information.
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constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information.
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static float FFT_MajorPeak = 0.0f;
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static float FFT_Magnitude = 0.0f;
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@ -274,9 +281,12 @@ void FFTcode(void * parameter)
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// Manual linear adjustment of gain using sampleGain adjustment for different input types.
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fftCalc[i] *= soundAgc ? multAgc : ((float)sampleGain/40.0f * (float)inputLevel/128.0f + 1.0f/16.0f); //with inputLevel adjustment
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// smooth results
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//fftAvg[i] = fftCalc[i]*0.05f + 0.95f*fftAvg[i]; // will need approx 10 cycles (250ms) for converging against fftCalc[i]
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fftAvg[i] = fftCalc[i] *0.1f + 0.9f*fftAvg[i]; // will need approx 5 cycles (125ms) for converging against fftCalc[i]
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// Now, let's dump it all into fftResult. Need to do this, otherwise other routines might grab fftResult values prematurely.
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fftResult[i] = constrain((int)fftCalc[i], 0, 254);
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fftAvg[i] = (float)fftResult[i]*0.05f + 0.95f*fftAvg[i];
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//fftResult[i] = constrain((int)fftCalc[i], 0, 254);
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fftResult[i] = constrain((int)fftAvg[i], 0, 254);
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}
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#ifdef WLED_DEBUG
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@ -602,10 +612,13 @@ class AudioReactive : public Usermod {
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// this is the minimal code for reading analog mic input on 8266.
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// warning!! Absolutely experimental code. Audio on 8266 is still not working. Expects a million follow-on problems.
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static unsigned long lastAnalogTime = 0;
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static float lastAnalogValue = 0.0f;
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if (millis() - lastAnalogTime > 20) {
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micDataReal = analogRead(A0); // read one sample with 10bit resolution. This is a dirty hack, supporting volumereactive effects only.
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lastAnalogTime = millis();
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}
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lastAnalogValue = micDataReal;
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yield();
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} else micDataReal = lastAnalogValue;
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micIn = int(micDataReal);
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#endif
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#endif
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@ -618,6 +631,7 @@ class AudioReactive : public Usermod {
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float micInNoDC = fabs(micDataReal - micLev);
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expAdjF = (weighting * micInNoDC + (1.0-weighting) * expAdjF);
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expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate
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if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0"
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expAdjF = fabsf(expAdjF); // Now (!) take the absolute value
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tmpSample = expAdjF;
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@ -664,14 +678,9 @@ class AudioReactive : public Usermod {
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/* Limits the dynamics of volumeSmth (= sampleAvg or sampleAgc).
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* It does not affect FFTResult[] or volumeRaw ( = sample or rawSampleAgc)
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* does not affect FFTResult[] or volumeRaw ( = sample or rawSampleAgc)
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*/
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// effects: Gravimeter, Gravcenter, Gravcentric, Noisefire, Plasmoid, Freqpixels, Freqwave, Gravfreq, (2D Swirl, 2D Waverly)
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// 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.
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// values below will be made user-configurable later
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const float attackTime = 200; // attack time -> 0.2sec
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const float decayTime = 2800; // decay time -> 2.8sec
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void limitSampleDynamics(void) {
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#ifdef SOUND_DYNAMICS_LIMITER
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const float bigChange = 196; // just a representative number - a large, expected sample value
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@ -681,8 +690,8 @@ class AudioReactive : public Usermod {
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if ((attackTime > 0) && (decayTime > 0)) { // only change volume if user has defined attack>0 and decay>0
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long delta_time = millis() - last_time;
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delta_time = constrain(delta_time , 1, 1000); // below 1ms -> 1ms; above 1sec -> sily lil hick-up
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float maxAttack = bigChange * float(delta_time) / attackTime;
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float maxDecay = - bigChange * float(delta_time) / decayTime;
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float maxAttack = bigChange * float(delta_time) / float(attackTime);
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float maxDecay = - bigChange * float(delta_time) / float(decayTime);
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float deltaSample = volumeSmth - last_volumeSmth;
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if (deltaSample > maxAttack) deltaSample = maxAttack;
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@ -704,8 +713,11 @@ class AudioReactive : public Usermod {
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audioSyncPacket transmitData;
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strncpy_P(transmitData.header, PSTR(UDP_SYNC_HEADER), 6);
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transmitData.sampleRaw = volumeRaw;
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transmitData.sampleSmth = volumeSmth;
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//transmitData.sampleRaw = volumeRaw;
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//transmitData.sampleSmth = volumeSmth;
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// transmit samples that were not modified by limitSampleDynamics()
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transmitData.sampleRaw = (soundAgc) ? rawSampleAgc: sampleRaw;
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transmitData.sampleSmth = (soundAgc) ? sampleAgc : sampleAvg;
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transmitData.samplePeak = udpSamplePeak ? 1:0;
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udpSamplePeak = false; // Reset udpSamplePeak after we've transmitted it
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transmitData.reserved1 = 0;
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@ -744,9 +756,11 @@ class AudioReactive : public Usermod {
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if (packetSize == sizeof(audioSyncPacket) && !(isValidUdpSyncVersion((const char *)fftBuff))) {
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audioSyncPacket *receivedPacket = reinterpret_cast<audioSyncPacket*>(fftBuff);
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// update samples for effects
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volumeSmth = receivedPacket->sampleSmth;
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volumeRaw = receivedPacket->sampleRaw;
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// update internal samples
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sampleRaw = volumeRaw;
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sampleAvg = volumeSmth;
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rawSampleAgc = volumeRaw;
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@ -945,6 +959,7 @@ class AudioReactive : public Usermod {
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if (audioSyncEnabled & 0x02) disableSoundProcessing = true; // make sure everything is disabled IF in audio Receive mode
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if (audioSyncEnabled & 0x01) disableSoundProcessing = false; // keep running audio IF we're in audio Transmit mode
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// Only run the sampling code IF we're not in Receive mode or realtime mode
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if (!(audioSyncEnabled & 0x02) && !disableSoundProcessing) {
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bool agcEffect = false;
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@ -981,9 +996,8 @@ class AudioReactive : public Usermod {
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limitSampleDynamics(); // optional - makes volumeSmth very smooth and fluent
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// update UI
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// update WebServer UI
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uint8_t knownMode = strip.getFirstSelectedSeg().mode; // 1st selected segment is more appropriate than main segment
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if (lastMode != knownMode) { // only execute if mode changes
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char lineBuffer[4];
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extractModeName(knownMode, JSON_mode_names, lineBuffer, 3); // use of JSON_mode_names is deprecated, use nullptr
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@ -1024,10 +1038,19 @@ class AudioReactive : public Usermod {
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}
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}
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// Begin UDP Microphone Sync
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if ((audioSyncEnabled & 0x02) && millis() - lastTime > delayMs) { // Only run the audio listener code if we're in Receive mode
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(void) receiveAudioData(); // ToDo: use return value for something meaningfull
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lastTime = millis();
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// UDP Microphone Sync - receive mode
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if ((audioSyncEnabled & 0x02) && udpSyncConnected) {
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// Only run the audio listener code if we're in Receive mode
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static float syncVolumeSmth = 0;
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bool have_new_sample = false;
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if (millis() - lastTime > delayMs) {
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have_new_sample = receiveAudioData();
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lastTime = millis();
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}
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if (have_new_sample) syncVolumeSmth = volumeSmth; // remember received sample
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else volumeSmth = syncVolumeSmth; // restore originally received sample for next run of dynamics limiter
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limitSampleDynamics(); // run dynamics limiter on received volumeSmth, to hide jumps and hickups
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}
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#if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
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@ -1036,12 +1059,13 @@ class AudioReactive : public Usermod {
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}
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#endif
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if ((audioSyncEnabled & 0x01) && millis() - lastTime > 20) { // Only run the transmit code IF we're in Transmit mode
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//UDP Microphone Sync - transmit mode
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if ((audioSyncEnabled & 0x01) && millis() - lastTime > 20) {
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// Only run the transmit code IF we're in Transmit mode
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transmitAudioData();
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lastTime = millis();
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}
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//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 :-(
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}
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@ -339,9 +339,12 @@ class I2SAdcSource : public I2SSource {
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#else
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.communication_format = i2s_comm_format_t(I2S_COMM_FORMAT_I2S | I2S_COMM_FORMAT_I2S_MSB),
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#endif
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.intr_alloc_flags = ESP_INTR_FLAG_LEVEL2,
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.intr_alloc_flags = ESP_INTR_FLAG_LEVEL1,
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.dma_buf_count = 8,
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.dma_buf_len = _blockSize
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.dma_buf_len = _blockSize,
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.use_apll = false,
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.tx_desc_auto_clear = false,
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.fixed_mclk = 0
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};
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}
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@ -350,6 +353,7 @@ class I2SAdcSource : public I2SSource {
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void initialize(int8_t audioPin, int8_t = I2S_PIN_NO_CHANGE, int8_t = I2S_PIN_NO_CHANGE, int8_t = I2S_PIN_NO_CHANGE, int8_t = I2S_PIN_NO_CHANGE, int8_t = I2S_PIN_NO_CHANGE) {
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if(!pinManager.allocatePin(audioPin, false, PinOwner::UM_Audioreactive)) {
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DEBUGSR_PRINTF("failed to allocate GPIO for audio analog input: %d\n", audioPin);
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return;
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}
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_audioPin = audioPin;
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@ -376,7 +380,7 @@ class I2SAdcSource : public I2SSource {
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DEBUGSR_PRINTF("Failed to set i2s adc mode: %d\n", err);
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return;
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}
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// adc1_config_channel_atten(adc1_channel_t(channel), ADC_ATTEN_DB_11)); //see https://github.com/espressif/arduino-esp32/blob/master/libraries/ESP32/examples/I2S/HiFreq_ADC/HiFreq_ADC.ino
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#if defined(ARDUINO_ARCH_ESP32)
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// according to docs from espressif, the ADC needs to be started explicitly
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// fingers crossed
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@ -408,7 +412,7 @@ class I2SAdcSource : public I2SSource {
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#if !defined(ARDUINO_ARCH_ESP32)
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// old code - works for me without enable/disable, at least on ESP32.
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err = i2s_adc_disable(I2S_NUM_0);
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err = i2s_adc_disable(I2S_NUM_0); //i2s_adc_disable() may cause crash with IDF 4.4 (https://github.com/espressif/arduino-esp32/issues/6832)
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//err = i2s_stop(I2S_NUM_0);
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if (err != ESP_OK) {
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DEBUGSR_PRINTF("Failed to disable i2s adc: %d\n", err);
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@ -6711,7 +6711,8 @@ uint16_t mode_freqmap(void) { // Map FFT_MajorPeak to SEGLEN.
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SEGMENT.fade_out(SEGMENT.speed);
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uint16_t locn = (log10f((float)FFT_MajorPeak) - 1.78f) * (float)SEGLEN/(3.71f-1.78f); // log10 frequency range is from 1.78 to 3.71. Let's scale to SEGLEN.
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int locn = (log10f((float)FFT_MajorPeak) - 1.78f) * (float)SEGLEN/(3.71f-1.78f); // log10 frequency range is from 1.78 to 3.71. Let's scale to SEGLEN.
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if (locn < 1) locn = 0; // avoid underflow
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if (locn >=SEGLEN) locn = SEGLEN-1;
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uint16_t pixCol = (log10f(FFT_MajorPeak) - 1.78f) * 255.0f/(3.71f-1.78f); // Scale log10 of frequency values to the 255 colour index.
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@ -216,10 +216,13 @@ void handleAnalog(uint8_t b)
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void handleButton()
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{
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static unsigned long lastRead = 0UL;
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static unsigned long lastRun = 0UL;
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bool analog = false;
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unsigned long now = millis();
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if (strip.isUpdating()) return; // don't interfere with strip updates. Our button will still be there in 1ms (next cycle)
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//if (strip.isUpdating()) return; // don't interfere with strip updates. Our button will still be there in 1ms (next cycle)
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if (strip.isUpdating() && (millis() - lastRun < 400)) return; // be niced, but avoid button starvation
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lastRun = millis();
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for (uint8_t b=0; b<WLED_MAX_BUTTONS; b++) {
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#ifdef ESP8266
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Ładowanie…
Reference in New Issue