lba.gitea/WLED
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

Addec config save/load.

Browse Source 
Changed double to float.
This commit is contained in:
Blaz Kristan 2022-06-09 18:55:35 +02:00
parent a6746f77f0
commit 1828a2a81c
1 changed files with 126 additions and 101 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

183
usermods/audioreactive/audio_reactive.h
View File

@ -39,8 +39,10 @@ constexpr int SAMPLE_RATE = 10240; // Base sample rate in Hz
//#define MAJORPEAK_SUPPRESS_NOISE // define to activate a dirty hack that ignores the lowest + hightest FFT bins //#define MAJORPEAK_SUPPRESS_NOISE // define to activate a dirty hack that ignores the lowest + hightest FFT bins
byte audioSyncEnabled = 0; // globals
uint16_t audioSyncPort = 11988; static byte audioSyncEnabled = 0;
static uint16_t audioSyncPort = 11988;
uint8_t inputLevel; // UI slider value uint8_t inputLevel; // UI slider value
// //
@ -154,7 +156,7 @@ void FFTcode(void * parameter) {
// taskYIELD(), yield(), vTaskDelay() and esp_task_wdt_feed() didn't seem to work. // taskYIELD(), yield(), vTaskDelay() and esp_task_wdt_feed() didn't seem to work.
// Only run the FFT computing code if we're not in Receive mode // Only run the FFT computing code if we're not in Receive mode
if (audioSyncEnabled & (1 << 1)) if (audioSyncEnabled & 0x02)
continue; continue;
audioSource->getSamples(vReal, samplesFFT); audioSource->getSamples(vReal, samplesFFT);
@ -293,29 +295,20 @@ void FFTcode(void * parameter) {
fftCalc[14] = (fftAdd(147,194)) /48; // 2940 - 3900 fftCalc[14] = (fftAdd(147,194)) /48; // 2940 - 3900
fftCalc[15] = (fftAdd(194, 255)) /62; // 3880 - 5120 fftCalc[15] = (fftAdd(194, 255)) /62; // 3880 - 5120
// Noise supression of fftCalc bins using soundSquelch adjustment for different input types.
for (int i=0; i < 16; i++) { for (int i=0; i < 16; i++) {
fftCalc[i] -= (float)soundSquelch*(float)linearNoise[i]/4.0 <= 0? 0 : fftCalc[i]; // Noise supression of fftCalc bins using soundSquelch adjustment for different input types.
} fftCalc[i] -= (float)soundSquelch * (float)linearNoise[i] / 4.0f <= 0.0f ? 0 : fftCalc[i];
// Adjustment for frequency curves. // Adjustment for frequency curves.
for (int i=0; i < 16; i++) {
fftCalc[i] *= fftResultPink[i]; fftCalc[i] *= fftResultPink[i];
}
// Manual linear adjustment of gain using sampleGain adjustment for different input types. // Manual linear adjustment of gain using sampleGain adjustment for different input types.
for (int i=0; i < 16; i++) { fftCalc[i] *= soundAgc ? multAgc : (float)sampleGain/40.0f * inputLevel/128 + (float)fftCalc[i]/16.0f; //with inputLevel adjustment
if (soundAgc)
fftCalc[i] = fftCalc[i] * multAgc;
else
fftCalc[i] = fftCalc[i] * (double)sampleGain / 40.0 * inputLevel/128 + (double)fftCalc[i]/16.0; //with inputLevel adjustment
}
// Now, let's dump it all into fftResult. Need to do this, otherwise other routines might grab fftResult values prematurely. // Now, let's dump it all into fftResult. Need to do this, otherwise other routines might grab fftResult values prematurely.
for (int i=0; i < 16; i++) {
// fftResult[i] = (int)fftCalc[i]; // fftResult[i] = (int)fftCalc[i];
fftResult[i] = constrain((int)fftCalc[i], 0, 254); // question: why do we constrain values to 8bit here ??? fftResult[i] = constrain((int)fftCalc[i], 0, 254); // question: why do we constrain values to 8bit here ???
fftAvg[i] = (float)fftResult[i]*.05 + (1-.05)*fftAvg[i]; fftAvg[i] = (float)fftResult[i]*0.05f + (1.0f - 0.05f)*fftAvg[i]; // why no just 0.95f*fftAvg[i]?
} }
// release second sample to volume reactive effects. // release second sample to volume reactive effects.
@ -364,7 +357,7 @@ class AudioReactive : public Usermod {
int8_t i2sckPin = I2S_CKPIN; int8_t i2sckPin = I2S_CKPIN;
#endif #endif
WiFiUDP fftUdp; #define UDP_SYNC_HEADER "00001"
struct audioSyncPacket { struct audioSyncPacket {
char header[6] = UDP_SYNC_HEADER; char header[6] = UDP_SYNC_HEADER;
uint8_t myVals[32]; // 32 Bytes uint8_t myVals[32]; // 32 Bytes
@ -377,8 +370,9 @@ class AudioReactive : public Usermod {
double FFT_MajorPeak; // 08 Bytes double FFT_MajorPeak; // 08 Bytes
}; };
WiFiUDP fftUdp;
// set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer) // set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer)
#define UDP_SYNC_HEADER "00001"
uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger
uint8_t binNum; // Used to select the bin for FFT based beat detection. uint8_t binNum; // Used to select the bin for FFT based beat detection.
@ -402,15 +396,16 @@ class AudioReactive : public Usermod {
float beat = 0.f; // beat Detection float beat = 0.f; // beat Detection
float expAdjF; // Used for exponential filter. float expAdjF; // Used for exponential filter.
float weighting = 0.2; // Exponential filter weighting. Will be adjustable in a future release. float weighting = 0.2f; // Exponential filter weighting. Will be adjustable in a future release.
bool udpSyncConnected = false;
// strings to reduce flash memory usage (used more than twice) // strings to reduce flash memory usage (used more than twice)
static const char _name[]; static const char _name[];
static const char _analogmic[];
static const char _digitalmic[];
double mapf(double x, double in_min, double in_max, double out_min, double out_max);
bool isValidUdpSyncVersion(char header[6]) { bool isValidUdpSyncVersion(char header[6]) {
if (strncmp(header, UDP_SYNC_HEADER, 6) == 0) { if (strncmp(header, UDP_SYNC_HEADER, 6) == 0) {
return true; return true;
@ -419,6 +414,7 @@ class AudioReactive : public Usermod {
} }
} }
void logAudio() { void logAudio() {
#ifdef MIC_LOGGER #ifdef MIC_LOGGER
//Serial.print("micData:"); Serial.print(micData); Serial.print("\t"); //Serial.print("micData:"); Serial.print(micData); Serial.print("\t");
@ -500,6 +496,7 @@ class AudioReactive : public Usermod {
#endif // FFT_SAMPLING_LOG #endif // FFT_SAMPLING_LOG
} // logAudio() } // logAudio()
/* /*
* A "PI control" multiplier to automatically adjust sound sensitivity. * A "PI control" multiplier to automatically adjust sound sensitivity.
* *
@ -520,10 +517,10 @@ class AudioReactive : public Usermod {
float tmpAgc = sampleReal * multAgc; // what-if amplified signal float tmpAgc = sampleReal * multAgc; // what-if amplified signal
float control_error; // "control error" input for PI control float control_error; // "control error" input for PI control
static double control_integrated = 0.0; // "integrator control" = accumulated error static float control_integrated = 0.0f; // "integrator control" = accumulated error
if (last_soundAgc != soundAgc) if (last_soundAgc != soundAgc)
control_integrated = 0.0; // new preset - reset integrator control_integrated = 0.0f; // new preset - reset integrator
// For PI control, we need to have a contant "frequency" // For PI control, we need to have a contant "frequency"
// so let's make sure that the control loop is not running at insane speed // so let's make sure that the control loop is not running at insane speed
@ -532,13 +529,13 @@ class AudioReactive : public Usermod {
if (time_now - last_time > 2) { if (time_now - last_time > 2) {
last_time = time_now; last_time = time_now;
if((fabs(sampleReal) < 2.0) || (sampleMax < 1.0)) { if((fabs(sampleReal) < 2.0f) || (sampleMax < 1.0f)) {
// MIC signal is "squelched" - deliver silence // MIC signal is "squelched" - deliver silence
multAgcTemp = multAgc; // keep old control value (no change) multAgcTemp = multAgc; // keep old control value (no change)
tmpAgc = 0; tmpAgc = 0;
// we need to "spin down" the intgrated error buffer // we need to "spin down" the intgrated error buffer
if (fabs(control_integrated) < 0.01) control_integrated = 0.0; if (fabs(control_integrated) < 0.01f) control_integrated = 0.0f;
else control_integrated = control_integrated * 0.91; else control_integrated = control_integrated * 0.91f;
} else { } else {
// compute new setpoint // compute new setpoint
if (tmpAgc <= agcTarget0Up[AGC_preset]) if (tmpAgc <= agcTarget0Up[AGC_preset])
@ -547,17 +544,17 @@ class AudioReactive : public Usermod {
multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint
} }
// limit amplification // limit amplification
if (multAgcTemp > 32.0) multAgcTemp = 32.0; if (multAgcTemp > 32.0f) multAgcTemp = 32.0f;
if (multAgcTemp < 1.0/64.0) multAgcTemp = 1.0/64.0; if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f;
// compute error terms // compute error terms
control_error = multAgcTemp - lastMultAgc; control_error = multAgcTemp - lastMultAgc;
if (((multAgcTemp > 0.085) && (multAgcTemp < 6.5)) //integrator anti-windup by clamping if (((multAgcTemp > 0.085f) && (multAgcTemp < 6.5f)) //integrator anti-windup by clamping
&& (multAgc*sampleMax < agcZoneStop[AGC_preset])) //integrator ceiling (>140% of max) && (multAgc*sampleMax < agcZoneStop[AGC_preset])) //integrator ceiling (>140% of max)
control_integrated += control_error * 0.002 * 0.25; // 2ms = intgration time; 0.25 for damping control_integrated += control_error * 0.002f * 0.25f; // 2ms = intgration time; 0.25 for damping
else else
control_integrated *= 0.9; // spin down that beasty integrator control_integrated *= 0.9f; // spin down that beasty integrator
// apply PI Control // apply PI Control
tmpAgc = sampleReal * lastMultAgc; // check "zone" of the signal using previous gain tmpAgc = sampleReal * lastMultAgc; // check "zone" of the signal using previous gain
@ -570,22 +567,22 @@ class AudioReactive : public Usermod {
} }
// limit amplification again - PI controler sometimes "overshoots" // limit amplification again - PI controler sometimes "overshoots"
if (multAgcTemp > 32.0) multAgcTemp = 32.0; if (multAgcTemp > 32.0f) multAgcTemp = 32.0f;
if (multAgcTemp < 1.0/64.0) multAgcTemp = 1.0/64.0; if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f;
} }
// NOW finally amplify the signal // NOW finally amplify the signal
tmpAgc = sampleReal * multAgcTemp; // apply gain to signal tmpAgc = sampleReal * multAgcTemp; // apply gain to signal
if(fabs(sampleReal) < 2.0) tmpAgc = 0; // apply squelch threshold if(fabs(sampleReal) < 2.0f) tmpAgc = 0; // apply squelch threshold
if (tmpAgc > 255) tmpAgc = 255; // limit to 8bit if (tmpAgc > 255) tmpAgc = 255; // limit to 8bit
if (tmpAgc < 1) tmpAgc = 0; // just to be sure if (tmpAgc < 1) tmpAgc = 0; // just to be sure
// update global vars ONCE - multAgc, sampleAGC, rawSampleAgc // update global vars ONCE - multAgc, sampleAGC, rawSampleAgc
multAgc = multAgcTemp; multAgc = multAgcTemp;
rawSampleAgc = 0.8 * tmpAgc + 0.2 * (float)rawSampleAgc; rawSampleAgc = 0.8f * tmpAgc + 0.2f * (float)rawSampleAgc;
// update smoothed AGC sample // update smoothed AGC sample
if(fabs(tmpAgc) < 1.0) if(fabs(tmpAgc) < 1.0f)
sampleAgc = 0.5 * tmpAgc + 0.5 * sampleAgc; // fast path to zero sampleAgc = 0.5f * tmpAgc + 0.5f * sampleAgc; // fast path to zero
else else
sampleAgc = sampleAgc + agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path sampleAgc = sampleAgc + agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
@ -595,6 +592,7 @@ class AudioReactive : public Usermod {
last_soundAgc = soundAgc; last_soundAgc = soundAgc;
} // agcAvg() } // agcAvg()
void getSample() { void getSample() {
static long peakTime; static long peakTime;
@ -617,8 +615,8 @@ class AudioReactive : public Usermod {
#endif #endif
// Note to self: the next line kills 80% of sample - "miclev" filter runs at "full arduino loop" speed, following the signal almost instantly! // 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 * 31) + micIn) / 32; // Smooth it out over the last 32 samples for automatic centering
micLev = ((micLev * 8191.0) + micDataReal) / 8192.0; // takes a few seconds to "catch up" with the Mic Input micLev = ((micLev * 8191.0f) + micDataReal) / 8192.0f; // takes a few seconds to "catch up" with the Mic Input
if(micIn < micLev) micLev = ((micLev * 31.0) + micDataReal) / 32.0; // align MicLev to lowest input signal 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 micIn -= micLev; // Let's center it to 0 now
/*---------DEBUG---------*/ /*---------DEBUG---------*/
@ -646,17 +644,17 @@ class AudioReactive : public Usermod {
sample = (int)sampleAdj; // ONLY update sample ONCE!!!! sample = (int)sampleAdj; // ONLY update sample ONCE!!!!
// keep "peak" sample, but decay value if current sample is below peak // keep "peak" sample, but decay value if current sample is below peak
if ((sampleMax < sampleReal) && (sampleReal > 0.5)) { if ((sampleMax < sampleReal) && (sampleReal > 0.5f)) {
sampleMax = sampleMax + 0.5 * (sampleReal - sampleMax); // new peak - with some filtering sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // new peak - with some filtering
} else { } else {
if ((multAgc*sampleMax > agcZoneStop[AGC_preset]) && (soundAgc > 0)) if ((multAgc*sampleMax > agcZoneStop[AGC_preset]) && (soundAgc > 0))
sampleMax = sampleMax + 0.5 * (sampleReal - sampleMax); // over AGC Zone - get back quickly sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // over AGC Zone - get back quickly
else else
sampleMax = sampleMax * agcSampleDecay[AGC_preset]; // signal to zero --> 5-8sec sampleMax = sampleMax * agcSampleDecay[AGC_preset]; // signal to zero --> 5-8sec
} }
if (sampleMax < 0.5) sampleMax = 0.0; if (sampleMax < 0.5f) sampleMax = 0.0f;
sampleAvg = ((sampleAvg * 15.0) + sampleAdj) / 16.0; // Smooth it out over the last 16 samples. sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
/*---------DEBUG---------*/ /*---------DEBUG---------*/
DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(sample); DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(sample);
@ -685,6 +683,7 @@ class AudioReactive : public Usermod {
} }
} // getSample() } // getSample()
/* /*
* A simple averaging multiplier to automatically adjust sound sensitivity. * A simple averaging multiplier to automatically adjust sound sensitivity.
*/ */
@ -696,17 +695,17 @@ class AudioReactive : public Usermod {
float lastMultAgc = multAgc; float lastMultAgc = multAgc;
float tmpAgc; float tmpAgc;
if(fabs(sampleAvg) < 2.0) { if(fabs(sampleAvg) < 2.0f) {
tmpAgc = sampleAvg; // signal below squelch -> deliver silence tmpAgc = sampleAvg; // signal below squelch -> deliver silence
multAgc = multAgc * 0.95; // slightly decrease gain multiplier multAgc = multAgc * 0.95f; // slightly decrease gain multiplier
} else { } else {
multAgc = (sampleAvg < 1) ? targetAgc : targetAgc / sampleAvg; // Make the multiplier so that sampleAvg * multiplier = setpoint multAgc = (sampleAvg < 1) ? targetAgc : targetAgc / sampleAvg; // Make the multiplier so that sampleAvg * multiplier = setpoint
} }
if (multAgc < 0.5) multAgc = 0.5; // signal higher than 2*setpoint -> don't reduce it further if (multAgc < 0.5f) multAgc = 0.5f; // signal higher than 2*setpoint -> don't reduce it further
multAgc = (lastMultAgc*127.0 +multAgc) / 128.0; //apply some filtering to gain multiplier -> smoother transitions multAgc = (lastMultAgc*127.0f +multAgc) / 128.0f; //apply some filtering to gain multiplier -> smoother transitions
tmpAgc = (float)sample * multAgc; // apply gain to signal tmpAgc = (float)sample * multAgc; // apply gain to signal
if (tmpAgc <= (soundSquelch*1.2)) tmpAgc = sample; // check against squelch threshold - increased by 20% to avoid artefacts (ripples) if (tmpAgc <= (soundSquelch*1.2f)) tmpAgc = sample; // check against squelch threshold - increased by 20% to avoid artefacts (ripples)
if (tmpAgc > 255) tmpAgc = 255; if (tmpAgc > 255) tmpAgc = 255;
sampleAgc = tmpAgc; // ONLY update sampleAgc ONCE because it's used elsewhere asynchronously!!!! sampleAgc = tmpAgc; // ONLY update sampleAgc ONCE because it's used elsewhere asynchronously!!!!
@ -714,6 +713,7 @@ class AudioReactive : public Usermod {
if (userVar0 > 255) userVar0 = 255; if (userVar0 > 255) userVar0 = 255;
} // agcAvg() } // agcAvg()
void transmitAudioData() { void transmitAudioData() {
if (!udpSyncConnected) return; if (!udpSyncConnected) return;
@ -852,7 +852,13 @@ class AudioReactive : public Usermod {
* Use it to initialize network interfaces * Use it to initialize network interfaces
*/ */
void connected() { void connected() {
//Serial.println("Connected to WiFi!"); if (audioSyncPort > 0 || (audioSyncEnabled & 0x03)) {
#ifndef ESP8266
udpSyncConnected = fftUdp.beginMulticast(IPAddress(239, 0, 0, 1), audioSyncPort);
#else
udpSyncConnected = fftUdp.beginMulticast(WiFi.localIP(), IPAddress(239, 0, 0, 1), audioSyncPort);
#endif
}
} }
@ -871,8 +877,7 @@ class AudioReactive : public Usermod {
lastTime = millis(); lastTime = millis();
} }
if (!(audioSyncEnabled & (1 << 1))) { // Only run the sampling code IF we're not in Receive mode if (!(audioSyncEnabled & 0x02)) { // Only run the sampling code IF we're not in Receive mode
lastTime = millis();
if (soundAgc > AGC_NUM_PRESETS) soundAgc = 0; // make sure that AGC preset is valid (to avoid array bounds violation) if (soundAgc > AGC_NUM_PRESETS) soundAgc = 0; // make sure that AGC preset is valid (to avoid array bounds violation)
getSample(); // Sample the microphone getSample(); // Sample the microphone
agcAvg(); // Calculated the PI adjusted value as sampleAvg agcAvg(); // Calculated the PI adjusted value as sampleAvg
@ -917,8 +922,8 @@ class AudioReactive : public Usermod {
last_kick_time = now_time; last_kick_time = now_time;
} }
int new_user_inputLevel = 128.0 * multAgc; // scale AGC multiplier so that "1" is at 128 int new_user_inputLevel = 128.0f * multAgc; // scale AGC multiplier so that "1" is at 128
if (multAgc > 1.0) new_user_inputLevel = 128.0 * (((multAgc - 1.0) / 4.0) +1.0); // compress range so we can show values up to 4 if (multAgc > 1.0f) new_user_inputLevel = 128.0f * (((multAgc - 1.0f) / 4.0f) +1.0f); // compress range so we can show values up to 4
new_user_inputLevel = MIN(MAX(new_user_inputLevel, 0),255); new_user_inputLevel = MIN(MAX(new_user_inputLevel, 0),255);
// update user interfaces - restrict frequency to avoid flooding UI's with small changes // update user interfaces - restrict frequency to avoid flooding UI's with small changes
@ -939,19 +944,17 @@ class AudioReactive : public Usermod {
logAudio(); logAudio();
} }
#endif #endif
} }
if (audioSyncEnabled & (1 << 0)) { // Only run the transmit code IF we're in Transmit mode
//Serial.println("Transmitting UDP Mic Packet");
if (audioSyncEnabled & 0x01) { // Only run the transmit code IF we're in Transmit mode
DEBUGSR_PRINTLN("Transmitting UDP Mic Packet");
EVERY_N_MILLIS(20) { EVERY_N_MILLIS(20) {
transmitAudioData(); transmitAudioData();
} }
} }
// Begin UDP Microphone Sync // Begin UDP Microphone Sync
if (audioSyncEnabled & (1 << 1)) { // Only run the audio listener code if we're in Receive mode if (audioSyncEnabled & 0x02) { // Only run the audio listener code if we're in Receive mode
if (millis()-lastTime > delayMs) { if (millis()-lastTime > delayMs) {
if (udpSyncConnected) { if (udpSyncConnected) {
//Serial.println("Checking for UDP Microphone Packet"); //Serial.println("Checking for UDP Microphone Packet");
@ -1003,7 +1006,6 @@ class AudioReactive : public Usermod {
} }
bool getUMData(um_data_t **data) { bool getUMData(um_data_t **data) {
if (!data) return false; // no pointer provided by caller -> exit if (!data) return false; // no pointer provided by caller -> exit
*data = &um_data; *data = &um_data;
@ -1042,6 +1044,7 @@ class AudioReactive : public Usermod {
} }
*/ */
/* /*
* readFromJsonState() can be used to receive data clients send to the /json/state part of the JSON API (state object). * readFromJsonState() can be used to receive data clients send to the /json/state part of the JSON API (state object).
* Values in the state object may be modified by connected clients * Values in the state object may be modified by connected clients
@ -1054,6 +1057,7 @@ class AudioReactive : public Usermod {
} }
*/ */
/* /*
* addToConfig() can be used to add custom persistent settings to the cfg.json file in the "um" (usermod) object. * addToConfig() can be used to add custom persistent settings to the cfg.json file in the "um" (usermod) object.
* It will be called by WLED when settings are actually saved (for example, LED settings are saved) * It will be called by WLED when settings are actually saved (for example, LED settings are saved)
@ -1092,10 +1096,26 @@ class AudioReactive : public Usermod {
void addToConfig(JsonObject& root) void addToConfig(JsonObject& root)
{ {
JsonObject top = root.createNestedObject(FPSTR(_name)); JsonObject top = root.createNestedObject(FPSTR(_name));
JsonArray pinArray = top.createNestedArray("pin");
pinArray.add(audioPin); JsonObject amic = top.createNestedObject(FPSTR(_analogmic));
top[F("audio_pin")] = audioPin; top["pin"] = audioPin;
//...
JsonObject dmic = top.createNestedObject(FPSTR(_digitalmic));
dmic[F("type")] = dmType;
JsonArray pinArray = dmic.createNestedArray("pin");
pinArray.add(i2ssdPin);
pinArray.add(i2swsPin);
pinArray.add(i2sckPin);
JsonObject cfg = top.createNestedObject("cfg");
cfg[F("squelch")] = soundSquelch;
cfg[F("gain")] = sampleGain;
cfg[F("AGC")] = soundAgc;
JsonObject sync = top.createNestedObject("sync");
sync[F("port")] = audioSyncPort;
sync[F("send")] = (bool) (audioSyncEnabled & 0x01);
sync[F("receive")] = (bool) (audioSyncEnabled & 0x02);
} }
@ -1116,26 +1136,29 @@ class AudioReactive : public Usermod {
*/ */
bool readFromConfig(JsonObject& root) bool readFromConfig(JsonObject& root)
{ {
// default settings values could be set here (or below using the 3-argument getJsonValue()) instead of in the class definition or constructor
// setting them inside readFromConfig() is slightly more robust, handling the rare but plausible use case of single value being missing after boot (e.g. if the cfg.json was manually edited and a value was removed)
JsonObject top = root[FPSTR(_name)]; JsonObject top = root[FPSTR(_name)];
bool configComplete = !top.isNull(); bool configComplete = !top.isNull();
configComplete &= getJsonValue(top[F("audio_pin")], audioPin); configComplete &= getJsonValue(top[FPSTR(_analogmic)]["pin"], audioPin);
/*
configComplete &= getJsonValue(top["testBool"], testBool); configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["type"], dmType);
configComplete &= getJsonValue(top["testULong"], testULong); configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][0], i2ssdPin);
configComplete &= getJsonValue(top["testFloat"], testFloat); configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][1], i2swsPin);
configComplete &= getJsonValue(top["testString"], testString); configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][2], i2sckPin);
configComplete &= getJsonValue(top["cfg"][F("squelch")], soundSquelch);
configComplete &= getJsonValue(top["cfg"][F("gain")], sampleGain);
configComplete &= getJsonValue(top["cfg"][F("AGC")], soundAgc);
configComplete &= getJsonValue(top["sync"][F("port")], audioSyncPort);
bool send = audioSyncEnabled & 0x01;
bool receive = audioSyncEnabled & 0x02;
configComplete &= getJsonValue(top["sync"][F("send")], send);
configComplete &= getJsonValue(top["sync"][F("receive")], receive);
audioSyncEnabled = send | (receive << 1);
// A 3-argument getJsonValue() assigns the 3rd argument as a default value if the Json value is missing
configComplete &= getJsonValue(top["testInt"], testInt, 42);
configComplete &= getJsonValue(top["testLong"], testLong, -42424242);
configComplete &= getJsonValue(top["pin"][0], testPins[0], -1);
configComplete &= getJsonValue(top["pin"][1], testPins[1], -1);
*/
return configComplete; return configComplete;
} }
@ -1163,3 +1186,5 @@ class AudioReactive : public Usermod {
// strings to reduce flash memory usage (used more than twice) // strings to reduce flash memory usage (used more than twice)
const char AudioReactive::_name[] PROGMEM = "AudioReactive"; const char AudioReactive::_name[] PROGMEM = "AudioReactive";
const char AudioReactive::_analogmic[] PROGMEM = "analogmic";
const char AudioReactive::_digitalmic[] PROGMEM = "digitalmic";