From cf54115077ed97bfa51fcf2c8653a6c876769652 Mon Sep 17 00:00:00 2001 From: Blaz Kristan Date: Thu, 16 Jun 2022 19:20:04 +0200 Subject: [PATCH] Sync bug fixes. Analog input fix. Code cleanup. --- usermods/audioreactive/audio_reactive.h | 192 ++++++++++-------------- usermods/audioreactive/audio_source.h | 2 +- 2 files changed, 81 insertions(+), 113 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index dc7a62af..2f9dfa8b 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -7,10 +7,6 @@ #error This audio reactive usermod does not support the ESP8266. #endif -//The SCL and SDA pins are defined here. -#define HW_PIN_SCL 22 -#define HW_PIN_SDA 21 - /* * Usermods allow you to add own functionality to WLED more easily * See: https://github.com/Aircoookie/WLED/wiki/Add-own-functionality @@ -170,18 +166,19 @@ void FFTcode(void * parameter) { const int halfSamplesFFT = samplesFFT / 2; // samplesFFT divided by 2 float maxSample1 = 0.0f; // max sample from first half of FFT batch float maxSample2 = 0.0f; // max sample from second half of FFT batch - for (int i=0; i < samplesFFT; i++) { + for (int i=0; i < halfSamplesFFT; i++) { // set imaginary parts to 0 vImag[i] = 0; // pick our our current mic sample - we take the max value from all samples that go into FFT if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) //skip extreme values - normally these are artefacts - { - if (i < halfSamplesFFT) { - if (fabsf((float)vReal[i]) > maxSample1) maxSample1 = fabsf((float)vReal[i]); - } else { - if (fabsf((float)vReal[i]) > maxSample2) maxSample2 = fabsf((float)vReal[i]); - } - } + if (fabsf((float)vReal[i]) > maxSample1) maxSample1 = fabsf((float)vReal[i]); + } + for (int i=halfSamplesFFT; i < samplesFFT; i++) { + // set imaginary parts to 0 + vImag[i] = 0; + // pick our our current mic sample - we take the max value from all samples that go into FFT + if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) //skip extreme values - normally these are artefacts + if (fabsf((float)vReal[i]) > maxSample2) maxSample2 = fabsf((float)vReal[i]); } // release first sample to volume reactive effects micDataSm = (uint16_t)maxSample1; @@ -315,11 +312,11 @@ void FFTcode(void * parameter) { #ifdef SR_DEBUG // Looking for fftResultMax for each bin using Pink Noise -// for (int i=0; i<16; i++) { -// fftResultMax[i] = ((fftResultMax[i] * 63.0) + fftResult[i]) / 64.0; -// Serial.print(fftResultMax[i]*fftResultPink[i]); Serial.print("\t"); -// } -// Serial.println(); + for (int i=0; i<16; i++) { + fftResultMax[i] = ((fftResultMax[i] * 63.0) + fftResult[i]) / 64.0; + DEBUGSR_PRINT(fftResultMax[i]*fftResultPink[i]); DEBUGSR_PRINT("\t"); + } + DEBUGSR_PRINTLN(); #endif } // for(;;) } // FFTcode() @@ -359,7 +356,7 @@ class AudioReactive : public Usermod { #else int8_t sdaPin = ES7243_SDAPIN; #endif - #ifndef ES7243_SDAPIN + #ifndef ES7243_SCLPIN int8_t sclPin = -1; #else int8_t sclPin = ES7243_SCLPIN; @@ -372,7 +369,7 @@ class AudioReactive : public Usermod { #define UDP_SYNC_HEADER "00001" struct audioSyncPacket { - char header[6] = UDP_SYNC_HEADER; + char header[6]; uint8_t myVals[32]; // 32 Bytes int sampleAgc; // 04 Bytes int sample; // 04 Bytes @@ -414,8 +411,11 @@ class AudioReactive : public Usermod { bool udpSyncConnected = false; + // used for AGC uint8_t lastMode = 0; // last known effect mode bool agcEffect = false; + int last_soundAgc = -1; + float control_integrated = 0.0f; // "integrator control" = accumulated error // strings to reduce flash memory usage (used more than twice) static const char _name[]; @@ -424,7 +424,7 @@ class AudioReactive : public Usermod { // private methods - bool isValidUdpSyncVersion(char header[6]) { + bool isValidUdpSyncVersion(const char *header) { return strncmp(header, UDP_SYNC_HEADER, 6) == 0; } @@ -524,14 +524,12 @@ class AudioReactive : public Usermod { */ void agcAvg() { const int AGC_preset = (soundAgc > 0)? (soundAgc-1): 0; // make sure the _compiler_ knows this value will not change while we are inside the function - static int last_soundAgc = -1; float lastMultAgc = multAgc; // last muliplier used float multAgcTemp = multAgc; // new multiplier float tmpAgc = sampleReal * multAgc; // what-if amplified signal float control_error; // "control error" input for PI control - static float control_integrated = 0.0f; // "integrator control" = accumulated error if (last_soundAgc != soundAgc) control_integrated = 0.0f; // new preset - reset integrator @@ -545,7 +543,7 @@ class AudioReactive : public Usermod { if((fabs(sampleReal) < 2.0f) || (sampleMax < 1.0f)) { // MIC signal is "squelched" - deliver silence - multAgcTemp = multAgc; // keep old control value (no change) + //multAgcTemp = multAgc; // keep old control value (no change) tmpAgc = 0; // we need to "spin down" the intgrated error buffer if (fabs(control_integrated) < 0.01f) control_integrated = 0.0f; @@ -617,27 +615,23 @@ class AudioReactive : public Usermod { micDataReal = micIn; #else micIn = micDataSm; // micDataSm = ((micData * 3) + micData)/4; - /*---------DEBUG---------*/ DEBUGSR_PRINT("micIn:\tmicData:\tmicIn>>2:\tmic_In_abs:\tsample:\tsampleAdj:\tsampleAvg:\n"); DEBUGSR_PRINT(micIn); DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(micData); - /*-------END DEBUG-------*/ - // We're still using 10 bit, but changing the analog read resolution in usermod.cpp - // if (digitalMic == false) micIn = micIn >> 2; // ESP32 has 2 more bits of A/D than ESP8266, so we need to normalize to 10 bit. - /*---------DEBUG---------*/ - DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(micIn); - /*-------END DEBUG-------*/ + + // We're still using 10 bit, but changing the analog read resolution in usermod.cpp + //if (digitalMic == false) micIn = micIn >> 2; // ESP32 has 2 more bits of A/D than ESP8266, so we need to normalize to 10 bit. + //DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(micIn); #endif + // Note to self: the next line kills 80% of sample - "miclev" filter runs at "full arduino loop" speed, following the signal almost instantly! //micLev = ((micLev * 31) + micIn) / 32; // Smooth it out over the last 32 samples for automatic centering micLev = ((micLev * 8191.0f) + micDataReal) / 8192.0f; // takes a few seconds to "catch up" with the Mic Input if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal micIn -= micLev; // Let's center it to 0 now - /*---------DEBUG---------*/ DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(micIn); - /*-------END DEBUG-------*/ - // Using an exponential filter to smooth out the signal. We'll add controls for this in a future release. + // Using an exponential filter to smooth out the signal. We'll add controls for this in a future release. float micInNoDC = fabs(micDataReal - micLev); expAdjF = (weighting * micInNoDC + (1.0-weighting) * expAdjF); expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate @@ -645,17 +639,16 @@ class AudioReactive : public Usermod { expAdjF = fabs(expAdjF); // Now (!) take the absolute value tmpSample = expAdjF; - /*---------DEBUG---------*/ DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(tmpSample); - /*-------END DEBUG-------*/ + micIn = abs(micIn); // And get the absolute value of each sample sampleAdj = tmpSample * sampleGain / 40 * inputLevel/128 + tmpSample / 16; // Adjust the gain. with inputLevel adjustment - // sampleReal = sampleAdj; + //sampleReal = sampleAdj; sampleReal = tmpSample; sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ??? - sample = (int)sampleAdj; // ONLY update sample ONCE!!!! + sample = (int16_t)sampleAdj; // ONLY update sample ONCE!!!! // keep "peak" sample, but decay value if current sample is below peak if ((sampleMax < sampleReal) && (sampleReal > 0.5f)) { @@ -670,10 +663,8 @@ class AudioReactive : public Usermod { sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples. - /*---------DEBUG---------*/ DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(sample); DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(sampleAvg); DEBUGSR_PRINT("\n\n"); - /*-------END DEBUG-------*/ // Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC uint16_t MinShowDelay = strip.getMinShowDelay(); @@ -700,18 +691,20 @@ class AudioReactive : public Usermod { void transmitAudioData() { if (!udpSyncConnected) return; + //DEBUGSR_PRINTLN("Transmitting UDP Mic Packet"); audioSyncPacket transmitData; + strncpy(transmitData.header, UDP_SYNC_HEADER, 6); for (int i = 0; i < 32; i++) { transmitData.myVals[i] = myVals[i]; } - transmitData.sampleAgc = sampleAgc; - transmitData.sample = sample; - transmitData.sampleAvg = sampleAvg; + transmitData.sampleAgc = sampleAgc; + transmitData.sample = sample; + transmitData.sampleAvg = sampleAvg; transmitData.samplePeak = udpSamplePeak; - udpSamplePeak = 0; // Reset udpSamplePeak after we've transmitted it + udpSamplePeak = 0; // Reset udpSamplePeak after we've transmitted it for (int i = 0; i < 16; i++) { transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254); @@ -727,6 +720,42 @@ class AudioReactive : public Usermod { } // transmitAudioData() + void receiveAudioData() { + if (!udpSyncConnected) return; + //DEBUGSR_PRINTLN("Checking for UDP Microphone Packet"); + + size_t packetSize = fftUdp.parsePacket(); + if (packetSize) { + //DEBUGSR_PRINTLN("Received UDP Sync Packet"); + uint8_t fftBuff[packetSize]; + fftUdp.read(fftBuff, packetSize); + + // VERIFY THAT THIS IS A COMPATIBLE PACKET + if (packetSize == sizeof(audioSyncPacket) && !(isValidUdpSyncVersion((const char *)fftBuff))) { + audioSyncPacket *receivedPacket = reinterpret_cast(fftBuff); + + for (int i = 0; i < 32; i++) myVals[i] = receivedPacket->myVals[i]; + + sampleAgc = receivedPacket->sampleAgc; + rawSampleAgc = receivedPacket->sampleAgc; + sample = receivedPacket->sample; + sampleAvg = receivedPacket->sampleAvg; + + // Only change samplePeak IF it's currently false. + // If it's true already, then the animation still needs to respond. + if (!samplePeak) samplePeak = receivedPacket->samplePeak; + + //These values are only available on the ESP32 + for (int i = 0; i < 16; i++) fftResult[i] = receivedPacket->fftResult[i]; + + FFT_Magnitude = receivedPacket->FFT_Magnitude; + FFT_MajorPeak = receivedPacket->FFT_MajorPeak; + //DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet"); + } + } + } + + public: //Functions called by WLED @@ -828,23 +857,10 @@ class AudioReactive : public Usermod { if (audioSource) audioSource->initialize(audioPin); break; } - - delay(250); - - //sampling_period_us = round(1000000*(1.0/SAMPLE_RATE)); + delay(250); // give mictophone enough time to initialise if (enabled) onUpdateBegin(false); // create FFT task -/* - // Define the FFT Task and lock it to core 0 - xTaskCreatePinnedToCore( - 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 - &FFT_Task, // Task handle - 0); // Core where the task should run -*/ + initDone = true; } @@ -885,8 +901,7 @@ class AudioReactive : public Usermod { if (lastMode != knownMode) { // only execute if mode changes char lineBuffer[3]; - /* uint8_t printedChars = */ extractModeName(knownMode, JSON_mode_names, lineBuffer, 3); //is this 'the' way to get mode name here? - + /* uint8_t printedChars = */ extractModeName(knownMode, JSON_mode_names, lineBuffer, 3); // use of JSON_mode_names is deprecated, use nullptr //used the following code to reverse engineer this // Serial.println(lineBuffer); // for (uint8_t i = 0; i 20) { - lastTime = millis(); + // Begin UDP Microphone Sync + if ((audioSyncEnabled & 0x02) && millis() - lastTime > delayMs) // Only run the audio listener code if we're in Receive mode + receiveAudioData(); + if (millis() - lastTime > 20) { if (audioSyncEnabled & 0x01) { // Only run the transmit code IF we're in Transmit mode - DEBUGSR_PRINTLN("Transmitting UDP Mic Packet"); transmitAudioData(); } - } - - // Begin UDP Microphone Sync - if ((audioSyncEnabled & 0x02) && udpSyncConnected) { // Only run the audio listener code if we're in Receive mode - if (millis()-lastTime > delayMs) { - //Serial.println("Checking for UDP Microphone Packet"); - int packetSize = fftUdp.parsePacket(); - if (packetSize) { - // Serial.println("Received UDP Sync Packet"); - uint8_t fftBuff[packetSize]; - fftUdp.read(fftBuff, packetSize); - audioSyncPacket receivedPacket; - memcpy(&receivedPacket, fftBuff, packetSize); - for (int i = 0; i < 32; i++ ){ - myVals[i] = receivedPacket.myVals[i]; - } - sampleAgc = receivedPacket.sampleAgc; - rawSampleAgc = receivedPacket.sampleAgc; - sample = receivedPacket.sample; - sampleAvg = receivedPacket.sampleAvg; - // VERIFY THAT THIS IS A COMPATIBLE PACKET - char packetHeader[6]; - memcpy(&receivedPacket, packetHeader, 6); - if (!(isValidUdpSyncVersion(packetHeader))) { - memcpy(&receivedPacket, fftBuff, packetSize); - for (int i = 0; i < 32; i++ ){ - myVals[i] = receivedPacket.myVals[i]; - } - sampleAgc = receivedPacket.sampleAgc; - rawSampleAgc = receivedPacket.sampleAgc; - sample = receivedPacket.sample; - sampleAvg = receivedPacket.sampleAvg; - - // Only change samplePeak IF it's currently false. - // If it's true already, then the animation still needs to respond. - if (!samplePeak) { - samplePeak = receivedPacket.samplePeak; - } - //These values are only available on the ESP32 - for (int i = 0; i < 16; i++) { - fftResult[i] = receivedPacket.fftResult[i]; - } - - FFT_Magnitude = receivedPacket.FFT_Magnitude; - FFT_MajorPeak = receivedPacket.FFT_MajorPeak; - //Serial.println("Finished parsing UDP Sync Packet"); - } - } - } + lastTime = millis(); } } diff --git a/usermods/audioreactive/audio_source.h b/usermods/audioreactive/audio_source.h index c8be6cb0..00c702c5 100644 --- a/usermods/audioreactive/audio_source.h +++ b/usermods/audioreactive/audio_source.h @@ -156,7 +156,7 @@ class I2SSource : public AudioSource { if (_mclkPin != I2S_PIN_NO_CHANGE) pinManager.deallocatePin(_mclkPin, PinOwner::UM_Audioreactive); } - void getSamples(double *buffer, uint16_t num_samples) { + virtual void getSamples(double *buffer, uint16_t num_samples) { if (_initialized) { esp_err_t err; size_t bytes_read = 0; /* Counter variable to check if we actually got enough data */