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AR: optimize sound sync, and code improvements

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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.
This commit is contained in:
Frank 2022-08-06 16:48:26 +02:00
parent a70717f2f7
commit 96d497a5cd
1 changed files with 83 additions and 48 deletions
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131
usermods/audioreactive/audio_reactive.h
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@ -58,12 +58,12 @@ static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1
// //
#define AGC_NUM_PRESETS 3 // AGC presets: normal, vivid, lazy #define AGC_NUM_PRESETS 3 // AGC presets: normal, vivid, lazy
const double agcSampleDecay[AGC_NUM_PRESETS] = { 0.9994f, 0.9985f, 0.9997f}; // decay factor for sampleMax, in case the current sample is below sampleMax const double agcSampleDecay[AGC_NUM_PRESETS] = { 0.9994f, 0.9985f, 0.9997f}; // decay factor for sampleMax, in case the current sample is below sampleMax
const float agcZoneLow[AGC_NUM_PRESETS] = { 32, 28, 36}; // low volume emergency zone const float agcZoneLow[AGC_NUM_PRESETS] = { 32, 28, 36}; // low volume emergency zone
const float agcZoneHigh[AGC_NUM_PRESETS] = { 240, 240, 248}; // high volume emergency zone const float agcZoneHigh[AGC_NUM_PRESETS] = { 240, 240, 248}; // high volume emergency zone
const float agcZoneStop[AGC_NUM_PRESETS] = { 336, 448, 304}; // disable AGC integrator if we get above this level const float agcZoneStop[AGC_NUM_PRESETS] = { 336, 448, 304}; // disable AGC integrator if we get above this level
const float agcTarget0[AGC_NUM_PRESETS] = { 112, 144, 164}; // first AGC setPoint -> between 40% and 65% const float agcTarget0[AGC_NUM_PRESETS] = { 112, 144, 164}; // first AGC setPoint -> between 40% and 65%
const float agcTarget0Up[AGC_NUM_PRESETS] = { 88, 64, 116}; // setpoint switching value (a poor man's bang-bang) const float agcTarget0Up[AGC_NUM_PRESETS] = { 88, 64, 116}; // setpoint switching value (a poor man's bang-bang)
const float agcTarget1[AGC_NUM_PRESETS] = { 220, 224, 216}; // second AGC setPoint -> around 85% const float agcTarget1[AGC_NUM_PRESETS] = { 220, 224, 216}; // second AGC setPoint -> around 85%
const double agcFollowFast[AGC_NUM_PRESETS] = { 1/192.f, 1/128.f, 1/256.f}; // quickly follow setpoint - ~0.15 sec const double agcFollowFast[AGC_NUM_PRESETS] = { 1/192.f, 1/128.f, 1/256.f}; // quickly follow setpoint - ~0.15 sec
const double agcFollowSlow[AGC_NUM_PRESETS] = {1/6144.f,1/4096.f,1/8192.f}; // slowly follow setpoint - ~2-15 secs const double agcFollowSlow[AGC_NUM_PRESETS] = {1/6144.f,1/4096.f,1/8192.f}; // slowly follow setpoint - ~2-15 secs
const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter
@ -308,7 +308,7 @@ void FFTcode(void * parameter)
fftCalc[ 4] = fftAddAvg(9,12); // 180 - 260 fftCalc[ 4] = fftAddAvg(9,12); // 180 - 260
fftCalc[ 5] = fftAddAvg(12,16); // 240 - 340 fftCalc[ 5] = fftAddAvg(12,16); // 240 - 340
fftCalc[ 6] = fftAddAvg(16,21); // 320 - 440 fftCalc[ 6] = fftAddAvg(16,21); // 320 - 440
fftCalc[ 7] = fftAddAvg(21,28); // 420 - 600 fftCalc[ 7] = fftAddAvg(21,29); // 420 - 600
fftCalc[ 8] = fftAddAvg(29,37); // 580 - 760 fftCalc[ 8] = fftAddAvg(29,37); // 580 - 760
fftCalc[ 9] = fftAddAvg(37,48); // 740 - 980 fftCalc[ 9] = fftAddAvg(37,48); // 740 - 980
fftCalc[10] = fftAddAvg(48,64); // 960 - 1300 fftCalc[10] = fftAddAvg(48,64); // 960 - 1300
@ -401,15 +401,29 @@ class AudioReactive : public Usermod {
int8_t mclkPin = MLCK_PIN; int8_t mclkPin = MLCK_PIN;
#endif #endif
// new "V2" audiosync struct - 40 Bytes
struct audioSyncPacket { struct audioSyncPacket {
char header[6]; char header[6]; // 06 Bytes
int sampleAgc; // 04 Bytes float sampleRaw; // 04 Bytes - either "sampleRaw" or "rawSampleAgc" depending on soundAgc setting
int sampleRaw; // 04 Bytes float sampleSmth; // 04 Bytes - either "sampleAvg" or "sampleAgc" depending on soundAgc setting
float sampleAvg; // 04 Bytes uint8_t samplePeak; // 01 Bytes - 0 no peak; >=1 peak detected. In future, this will also provide peak Magnitude
bool samplePeak; // 01 Bytes uint8_t reserved1; // 01 Bytes - for future extensions - not used yet
uint8_t fftResult[16]; // 16 Bytes uint8_t fftResult[16]; // 16 Bytes
double FFT_Magnitude; // 08 Bytes float FFT_Magnitude; // 04 Bytes
double FFT_MajorPeak; // 08 Bytes float FFT_MajorPeak; // 04 Bytes
};
// old "V1" audiosync struct - 83 Bytes - for backwards compatibility
struct audioSyncPacket_v1 {
char header[6]; // 06 Bytes
uint8_t myVals[32]; // 32 Bytes
int sampleAgc; // 04 Bytes
int sampleRaw; // 04 Bytes
float sampleAvg; // 04 Bytes
bool samplePeak; // 01 Bytes
uint8_t fftResult[16]; // 16 Bytes
double FFT_Magnitude; // 08 Bytes
double FFT_MajorPeak; // 08 Bytes
}; };
WiFiUDP fftUdp; WiFiUDP fftUdp;
@ -460,6 +474,7 @@ class AudioReactive : public Usermod {
static const char _analogmic[]; static const char _analogmic[];
static const char _digitalmic[]; static const char _digitalmic[];
static const char UDP_SYNC_HEADER[]; static const char UDP_SYNC_HEADER[];
static const char UDP_SYNC_HEADER_v1[];
float my_magnitude; float my_magnitude;
@ -642,14 +657,6 @@ class AudioReactive : public Usermod {
//if (userVar0 > 255) userVar0 = 255; //if (userVar0 > 255) userVar0 = 255;
last_soundAgc = soundAgc; last_soundAgc = soundAgc;
volumeSmth = (soundAgc) ? sampleAgc:sampleAvg;
volumeRaw = (soundAgc) ? rawSampleAgc : sampleRaw;
my_magnitude = FFT_Magnitude; // / 16.0f, 8.0f, 4.0f done in effects
if (soundAgc) my_magnitude *= multAgc;
if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
} // agcAvg() } // agcAvg()
@ -749,17 +756,17 @@ class AudioReactive : public Usermod {
audioSyncPacket transmitData; audioSyncPacket transmitData;
strncpy_P(transmitData.header, PSTR(UDP_SYNC_HEADER), 6); strncpy_P(transmitData.header, PSTR(UDP_SYNC_HEADER), 6);
transmitData.sampleAgc = sampleAgc; transmitData.sampleRaw = volumeRaw;
transmitData.sampleRaw = sampleRaw; transmitData.sampleSmth = volumeSmth;
transmitData.sampleAvg = sampleAvg; transmitData.samplePeak = udpSamplePeak ? 1:0;
transmitData.samplePeak = udpSamplePeak; udpSamplePeak = false; // Reset udpSamplePeak after we've transmitted it
udpSamplePeak = 0; // Reset udpSamplePeak after we've transmitted it transmitData.reserved1 = 0;
for (int i = 0; i < 16; i++) { for (int i = 0; i < 16; i++) {
transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254); transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254);
} }
transmitData.FFT_Magnitude = FFT_Magnitude; transmitData.FFT_Magnitude = my_magnitude;
transmitData.FFT_MajorPeak = FFT_MajorPeak; transmitData.FFT_MajorPeak = FFT_MajorPeak;
fftUdp.beginMulticastPacket(); fftUdp.beginMulticastPacket();
@ -780,7 +787,7 @@ class AudioReactive : public Usermod {
//DEBUGSR_PRINTLN("Checking for UDP Microphone Packet"); //DEBUGSR_PRINTLN("Checking for UDP Microphone Packet");
size_t packetSize = fftUdp.parsePacket(); size_t packetSize = fftUdp.parsePacket();
if (packetSize) { if (packetSize > 5) {
//DEBUGSR_PRINTLN("Received UDP Sync Packet"); //DEBUGSR_PRINTLN("Received UDP Sync Packet");
uint8_t fftBuff[packetSize]; uint8_t fftBuff[packetSize];
fftUdp.read(fftBuff, packetSize); fftUdp.read(fftBuff, packetSize);
@ -789,19 +796,35 @@ class AudioReactive : public Usermod {
if (packetSize == sizeof(audioSyncPacket) && !(isValidUdpSyncVersion((const char *)fftBuff))) { if (packetSize == sizeof(audioSyncPacket) && !(isValidUdpSyncVersion((const char *)fftBuff))) {
audioSyncPacket *receivedPacket = reinterpret_cast<audioSyncPacket*>(fftBuff); audioSyncPacket *receivedPacket = reinterpret_cast<audioSyncPacket*>(fftBuff);
sampleAgc = receivedPacket->sampleAgc; volumeSmth = receivedPacket->sampleSmth;
rawSampleAgc = receivedPacket->sampleAgc; volumeRaw = receivedPacket->sampleRaw;
sampleRaw = receivedPacket->sampleRaw;
sampleAvg = receivedPacket->sampleAvg;
sampleRaw = volumeRaw;
sampleAvg = volumeSmth;
rawSampleAgc = volumeRaw;
sampleAgc = volumeSmth;
multAgc = 1.0f;
// auto-reset sample peak. Need to do it here, because getSample() is not running
uint16_t MinShowDelay = strip.getMinShowDelay();
if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
samplePeak = false;
udpSamplePeak = false;
}
//if (userVar1 == 0) samplePeak = 0;
// Only change samplePeak IF it's currently false. // Only change samplePeak IF it's currently false.
// If it's true already, then the animation still needs to respond. // If it's true already, then the animation still needs to respond.
if (!samplePeak) samplePeak = receivedPacket->samplePeak; if (!samplePeak) {
samplePeak = receivedPacket->samplePeak >0 ? true:false;
if (samplePeak) timeOfPeak = millis();
//userVar1 = samplePeak;
}
//These values are only available on the ESP32 //These values are only available on the ESP32
for (int i = 0; i < 16; i++) fftResult[i] = receivedPacket->fftResult[i]; for (int i = 0; i < 16; i++) fftResult[i] = receivedPacket->fftResult[i];
FFT_Magnitude = receivedPacket->FFT_Magnitude; my_magnitude = receivedPacket->FFT_Magnitude;
FFT_Magnitude = my_magnitude;
FFT_MajorPeak = receivedPacket->FFT_MajorPeak; FFT_MajorPeak = receivedPacket->FFT_MajorPeak;
//DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet"); //DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet");
} }
@ -938,7 +961,7 @@ class AudioReactive : public Usermod {
} }
// We cannot wait indefinitely before processing audio data // We cannot wait indefinitely before processing audio data
//if (!enabled || strip.isUpdating()) return; //if (!enabled || strip.isUpdating()) return;
if (strip.isUpdating() && (millis() - lastUMRun < 12)) return; // be nice, but not too nice if (strip.isUpdating() && (millis() - lastUMRun < 2)) return; // be nice, but not too nice
// suspend local sound processing when "real time mode" is active (E131, UDP, ADALIGHT, ARTNET) // suspend local sound processing when "real time mode" is active (E131, UDP, ADALIGHT, ARTNET)
if ( (realtimeOverride == REALTIME_OVERRIDE_NONE) // please odd other orrides here if needed if ( (realtimeOverride == REALTIME_OVERRIDE_NONE) // please odd other orrides here if needed
@ -992,6 +1015,17 @@ class AudioReactive : public Usermod {
getSample(); // Sample the microphone getSample(); // Sample the microphone
agcAvg(); // Calculated the PI adjusted value as sampleAvg agcAvg(); // Calculated the PI adjusted value as sampleAvg
// update samples for effects (raw, smooth)
volumeSmth = (soundAgc) ? sampleAgc : sampleAvg;
volumeRaw = (soundAgc) ? rawSampleAgc: sampleRaw;
// update FFTMagnitude, taking into account AGC amplification
my_magnitude = FFT_Magnitude; // / 16.0f, 8.0f, 4.0f done in effects
if (soundAgc) my_magnitude *= multAgc;
if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
// update UI
uint8_t knownMode = strip.getFirstSelectedSeg().mode; // 1st selected segment is more appropriate than main segment uint8_t knownMode = strip.getFirstSelectedSeg().mode; // 1st selected segment is more appropriate than main segment
if (lastMode != knownMode) { // only execute if mode changes if (lastMode != knownMode) { // only execute if mode changes
@ -1032,22 +1066,22 @@ class AudioReactive : public Usermod {
last_user_inputLevel = new_user_inputLevel; last_user_inputLevel = new_user_inputLevel;
} }
} }
#if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
EVERY_N_MILLIS(20) {
logAudio();
}
#endif
} }
// Begin UDP Microphone Sync // Begin UDP Microphone Sync
if ((audioSyncEnabled & 0x02) && millis() - lastTime > delayMs) // Only run the audio listener code if we're in Receive mode if ((audioSyncEnabled & 0x02) && millis() - lastTime > delayMs) { // Only run the audio listener code if we're in Receive mode
receiveAudioData(); receiveAudioData();
lastTime = millis();
}
if (millis() - lastTime > 20) { #if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
if (audioSyncEnabled & 0x01) { // Only run the transmit code IF we're in Transmit mode EVERY_N_MILLIS(20) {
transmitAudioData(); logAudio();
} }
#endif
if ((audioSyncEnabled & 0x01) && millis() - lastTime > 20) { // Only run the transmit code IF we're in Transmit mode
transmitAudioData();
lastTime = millis(); lastTime = millis();
} }
} }
@ -1350,4 +1384,5 @@ const char AudioReactive::_enabled[] PROGMEM = "enabled";
const char AudioReactive::_inputLvl[] PROGMEM = "inputLevel"; const char AudioReactive::_inputLvl[] PROGMEM = "inputLevel";
const char AudioReactive::_analogmic[] PROGMEM = "analogmic"; const char AudioReactive::_analogmic[] PROGMEM = "analogmic";
const char AudioReactive::_digitalmic[] PROGMEM = "digitalmic"; const char AudioReactive::_digitalmic[] PROGMEM = "digitalmic";
const char AudioReactive::UDP_SYNC_HEADER[] PROGMEM = "00001"; 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