AR: optimize sound sync, and code improvements
UDP audio sync: introduced new header version, because the new struct (without myvals[]) is not compatible with the previous struct. Also optimized structure size. UDP audio sync: sender decides is AGC or non-AGC samples are transmitted. getsamples: move volumeSmth/volumeRaw code out of AGC core function.
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Frank
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@ -58,12 +58,12 @@ static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1
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//
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#define AGC_NUM_PRESETS 3 // AGC presets: normal, vivid, lazy
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const double agcSampleDecay[AGC_NUM_PRESETS] = { 0.9994f, 0.9985f, 0.9997f}; // decay factor for sampleMax, in case the current sample is below sampleMax
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const float agcZoneLow[AGC_NUM_PRESETS] = { 32, 28, 36}; // low volume emergency zone
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const float agcZoneHigh[AGC_NUM_PRESETS] = { 240, 240, 248}; // high volume emergency zone
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const float agcZoneStop[AGC_NUM_PRESETS] = { 336, 448, 304}; // disable AGC integrator if we get above this level
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const float agcTarget0[AGC_NUM_PRESETS] = { 112, 144, 164}; // first AGC setPoint -> between 40% and 65%
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const float agcTarget0Up[AGC_NUM_PRESETS] = { 88, 64, 116}; // setpoint switching value (a poor man's bang-bang)
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const float agcTarget1[AGC_NUM_PRESETS] = { 220, 224, 216}; // second AGC setPoint -> around 85%
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const float agcZoneLow[AGC_NUM_PRESETS] = { 32, 28, 36}; // low volume emergency zone
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const float agcZoneHigh[AGC_NUM_PRESETS] = { 240, 240, 248}; // high volume emergency zone
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const float agcZoneStop[AGC_NUM_PRESETS] = { 336, 448, 304}; // disable AGC integrator if we get above this level
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const float agcTarget0[AGC_NUM_PRESETS] = { 112, 144, 164}; // first AGC setPoint -> between 40% and 65%
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const float agcTarget0Up[AGC_NUM_PRESETS] = { 88, 64, 116}; // setpoint switching value (a poor man's bang-bang)
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const float agcTarget1[AGC_NUM_PRESETS] = { 220, 224, 216}; // second AGC setPoint -> around 85%
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const double agcFollowFast[AGC_NUM_PRESETS] = { 1/192.f, 1/128.f, 1/256.f}; // quickly follow setpoint - ~0.15 sec
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const double agcFollowSlow[AGC_NUM_PRESETS] = {1/6144.f,1/4096.f,1/8192.f}; // slowly follow setpoint - ~2-15 secs
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const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter
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@ -308,7 +308,7 @@ void FFTcode(void * parameter)
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fftCalc[ 4] = fftAddAvg(9,12); // 180 - 260
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fftCalc[ 5] = fftAddAvg(12,16); // 240 - 340
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fftCalc[ 6] = fftAddAvg(16,21); // 320 - 440
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fftCalc[ 7] = fftAddAvg(21,28); // 420 - 600
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fftCalc[ 7] = fftAddAvg(21,29); // 420 - 600
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fftCalc[ 8] = fftAddAvg(29,37); // 580 - 760
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fftCalc[ 9] = fftAddAvg(37,48); // 740 - 980
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fftCalc[10] = fftAddAvg(48,64); // 960 - 1300
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@ -401,15 +401,29 @@ class AudioReactive : public Usermod {
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int8_t mclkPin = MLCK_PIN;
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#endif
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// new "V2" audiosync struct - 40 Bytes
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struct audioSyncPacket {
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char header[6];
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int sampleAgc; // 04 Bytes
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int sampleRaw; // 04 Bytes
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float sampleAvg; // 04 Bytes
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bool samplePeak; // 01 Bytes
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char header[6]; // 06 Bytes
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float sampleRaw; // 04 Bytes - either "sampleRaw" or "rawSampleAgc" depending on soundAgc setting
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float sampleSmth; // 04 Bytes - either "sampleAvg" or "sampleAgc" depending on soundAgc setting
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uint8_t samplePeak; // 01 Bytes - 0 no peak; >=1 peak detected. In future, this will also provide peak Magnitude
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uint8_t reserved1; // 01 Bytes - for future extensions - not used yet
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uint8_t fftResult[16]; // 16 Bytes
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double FFT_Magnitude; // 08 Bytes
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double FFT_MajorPeak; // 08 Bytes
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float FFT_Magnitude; // 04 Bytes
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float FFT_MajorPeak; // 04 Bytes
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};
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// old "V1" audiosync struct - 83 Bytes - for backwards compatibility
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struct audioSyncPacket_v1 {
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char header[6]; // 06 Bytes
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uint8_t myVals[32]; // 32 Bytes
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int sampleAgc; // 04 Bytes
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int sampleRaw; // 04 Bytes
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float sampleAvg; // 04 Bytes
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bool samplePeak; // 01 Bytes
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uint8_t fftResult[16]; // 16 Bytes
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double FFT_Magnitude; // 08 Bytes
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double FFT_MajorPeak; // 08 Bytes
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};
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WiFiUDP fftUdp;
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@ -460,6 +474,7 @@ class AudioReactive : public Usermod {
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static const char _analogmic[];
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static const char _digitalmic[];
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static const char UDP_SYNC_HEADER[];
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static const char UDP_SYNC_HEADER_v1[];
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float my_magnitude;
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@ -642,14 +657,6 @@ class AudioReactive : public Usermod {
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//if (userVar0 > 255) userVar0 = 255;
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last_soundAgc = soundAgc;
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volumeSmth = (soundAgc) ? sampleAgc:sampleAvg;
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volumeRaw = (soundAgc) ? rawSampleAgc : sampleRaw;
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my_magnitude = FFT_Magnitude; // / 16.0f, 8.0f, 4.0f done in effects
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if (soundAgc) my_magnitude *= multAgc;
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if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
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} // agcAvg()
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@ -749,17 +756,17 @@ 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.sampleAgc = sampleAgc;
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transmitData.sampleRaw = sampleRaw;
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transmitData.sampleAvg = sampleAvg;
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transmitData.samplePeak = udpSamplePeak;
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udpSamplePeak = 0; // Reset udpSamplePeak after we've transmitted it
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transmitData.sampleRaw = volumeRaw;
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transmitData.sampleSmth = volumeSmth;
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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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for (int i = 0; i < 16; i++) {
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transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254);
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}
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transmitData.FFT_Magnitude = FFT_Magnitude;
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transmitData.FFT_Magnitude = my_magnitude;
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transmitData.FFT_MajorPeak = FFT_MajorPeak;
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fftUdp.beginMulticastPacket();
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@ -780,7 +787,7 @@ class AudioReactive : public Usermod {
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//DEBUGSR_PRINTLN("Checking for UDP Microphone Packet");
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size_t packetSize = fftUdp.parsePacket();
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if (packetSize) {
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if (packetSize > 5) {
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//DEBUGSR_PRINTLN("Received UDP Sync Packet");
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uint8_t fftBuff[packetSize];
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fftUdp.read(fftBuff, packetSize);
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@ -789,19 +796,35 @@ 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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sampleAgc = receivedPacket->sampleAgc;
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rawSampleAgc = receivedPacket->sampleAgc;
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sampleRaw = receivedPacket->sampleRaw;
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sampleAvg = receivedPacket->sampleAvg;
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volumeSmth = receivedPacket->sampleSmth;
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volumeRaw = receivedPacket->sampleRaw;
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sampleRaw = volumeRaw;
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sampleAvg = volumeSmth;
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rawSampleAgc = volumeRaw;
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sampleAgc = volumeSmth;
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multAgc = 1.0f;
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// auto-reset sample peak. Need to do it here, because getSample() is not running
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uint16_t MinShowDelay = strip.getMinShowDelay();
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if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
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samplePeak = false;
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udpSamplePeak = false;
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}
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//if (userVar1 == 0) samplePeak = 0;
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// Only change samplePeak IF it's currently false.
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// If it's true already, then the animation still needs to respond.
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if (!samplePeak) samplePeak = receivedPacket->samplePeak;
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if (!samplePeak) {
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samplePeak = receivedPacket->samplePeak >0 ? true:false;
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if (samplePeak) timeOfPeak = millis();
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//userVar1 = samplePeak;
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}
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//These values are only available on the ESP32
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for (int i = 0; i < 16; i++) fftResult[i] = receivedPacket->fftResult[i];
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FFT_Magnitude = receivedPacket->FFT_Magnitude;
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my_magnitude = receivedPacket->FFT_Magnitude;
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FFT_Magnitude = my_magnitude;
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FFT_MajorPeak = receivedPacket->FFT_MajorPeak;
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//DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet");
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}
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@ -938,7 +961,7 @@ class AudioReactive : public Usermod {
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}
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// We cannot wait indefinitely before processing audio data
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//if (!enabled || strip.isUpdating()) return;
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if (strip.isUpdating() && (millis() - lastUMRun < 12)) return; // be nice, but not too nice
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if (strip.isUpdating() && (millis() - lastUMRun < 2)) return; // be nice, but not too nice
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// suspend local sound processing when "real time mode" is active (E131, UDP, ADALIGHT, ARTNET)
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if ( (realtimeOverride == REALTIME_OVERRIDE_NONE) // please odd other orrides here if needed
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@ -992,6 +1015,17 @@ class AudioReactive : public Usermod {
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getSample(); // Sample the microphone
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agcAvg(); // Calculated the PI adjusted value as sampleAvg
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// update samples for effects (raw, smooth)
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volumeSmth = (soundAgc) ? sampleAgc : sampleAvg;
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volumeRaw = (soundAgc) ? rawSampleAgc: sampleRaw;
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// update FFTMagnitude, taking into account AGC amplification
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my_magnitude = FFT_Magnitude; // / 16.0f, 8.0f, 4.0f done in effects
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if (soundAgc) my_magnitude *= multAgc;
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if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
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// update 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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@ -1032,22 +1066,22 @@ class AudioReactive : public Usermod {
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last_user_inputLevel = new_user_inputLevel;
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}
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}
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#if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
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EVERY_N_MILLIS(20) {
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logAudio();
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}
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#endif
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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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if ((audioSyncEnabled & 0x02) && millis() - lastTime > delayMs) { // Only run the audio listener code if we're in Receive mode
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receiveAudioData();
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lastTime = millis();
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}
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if (millis() - lastTime > 20) {
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if (audioSyncEnabled & 0x01) { // Only run the transmit code IF we're in Transmit mode
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transmitAudioData();
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}
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#if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
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EVERY_N_MILLIS(20) {
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logAudio();
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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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transmitAudioData();
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lastTime = millis();
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}
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}
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@ -1350,4 +1384,5 @@ const char AudioReactive::_enabled[] PROGMEM = "enabled";
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const char AudioReactive::_inputLvl[] PROGMEM = "inputLevel";
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const char AudioReactive::_analogmic[] PROGMEM = "analogmic";
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const char AudioReactive::_digitalmic[] PROGMEM = "digitalmic";
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const char AudioReactive::UDP_SYNC_HEADER[] PROGMEM = "00001";
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const char AudioReactive::UDP_SYNC_HEADER[] PROGMEM = "00002"; // new sync header version, as format no longer compatible with previous structure
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const char AudioReactive::UDP_SYNC_HEADER_v1[] PROGMEM = "00001"; // old sync header version - need to add backwards-compatibility feature
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