Addec config save/load.
Changed double to float.
This commit is contained in:
Blaz Kristan
parent
a6746f77f0
commit
1828a2a81c
@ -39,8 +39,10 @@ constexpr int SAMPLE_RATE = 10240; // Base sample rate in Hz
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//#define MAJORPEAK_SUPPRESS_NOISE // define to activate a dirty hack that ignores the lowest + hightest FFT bins
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byte audioSyncEnabled = 0;
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uint16_t audioSyncPort = 11988;
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// globals
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static byte audioSyncEnabled = 0;
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static uint16_t audioSyncPort = 11988;
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uint8_t inputLevel; // UI slider value
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//
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@ -154,7 +156,7 @@ void FFTcode(void * parameter) {
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// taskYIELD(), yield(), vTaskDelay() and esp_task_wdt_feed() didn't seem to work.
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// Only run the FFT computing code if we're not in Receive mode
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if (audioSyncEnabled & (1 << 1))
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if (audioSyncEnabled & 0x02)
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continue;
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audioSource->getSamples(vReal, samplesFFT);
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@ -174,11 +176,11 @@ void FFTcode(void * parameter) {
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// pick our our current mic sample - we take the max value from all samples that go into FFT
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if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) //skip extreme values - normally these are artefacts
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{
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if (i <= halfSamplesFFT) {
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if (fabs(vReal[i]) > maxSample1) maxSample1 = fabs(vReal[i]);
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} else {
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if (fabs(vReal[i]) > maxSample2) maxSample2 = fabs(vReal[i]);
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}
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if (i <= halfSamplesFFT) {
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if (fabs(vReal[i]) > maxSample1) maxSample1 = fabs(vReal[i]);
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} else {
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if (fabs(vReal[i]) > maxSample2) maxSample2 = fabs(vReal[i]);
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}
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}
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}
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// release first sample to volume reactive effects
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@ -190,7 +192,7 @@ void FFTcode(void * parameter) {
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//FFT.Windowing( FFT_WIN_TYP_HAMMING, FFT_FORWARD ); // Weigh data - standard Hamming window
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//FFT.Windowing( FFT_WIN_TYP_BLACKMAN, FFT_FORWARD ); // Blackman window - better side freq rejection
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//FFT.Windowing( FFT_WIN_TYP_BLACKMAN_HARRIS, FFT_FORWARD );// Blackman-Harris - excellent sideband rejection
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FFT.Windowing( FFT_WIN_TYP_FLT_TOP, FFT_FORWARD ); // Flat Top Window - better amplitude accuracy
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FFT.Windowing( FFT_WIN_TYP_FLT_TOP, FFT_FORWARD ); // Flat Top Window - better amplitude accuracy
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FFT.Compute( FFT_FORWARD ); // Compute FFT
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FFT.ComplexToMagnitude(); // Compute magnitudes
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@ -293,29 +295,20 @@ void FFTcode(void * parameter) {
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fftCalc[14] = (fftAdd(147,194)) /48; // 2940 - 3900
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fftCalc[15] = (fftAdd(194, 255)) /62; // 3880 - 5120
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// Noise supression of fftCalc bins using soundSquelch adjustment for different input types.
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for (int i=0; i < 16; i++) {
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fftCalc[i] -= (float)soundSquelch*(float)linearNoise[i]/4.0 <= 0? 0 : fftCalc[i];
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}
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// Adjustment for frequency curves.
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for (int i=0; i < 16; i++) {
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// Noise supression of fftCalc bins using soundSquelch adjustment for different input types.
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fftCalc[i] -= (float)soundSquelch * (float)linearNoise[i] / 4.0f <= 0.0f ? 0 : fftCalc[i];
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// Adjustment for frequency curves.
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fftCalc[i] *= fftResultPink[i];
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}
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// Manual linear adjustment of gain using sampleGain adjustment for different input types.
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for (int i=0; i < 16; i++) {
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if (soundAgc)
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fftCalc[i] = fftCalc[i] * multAgc;
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else
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fftCalc[i] = fftCalc[i] * (double)sampleGain / 40.0 * inputLevel/128 + (double)fftCalc[i]/16.0; //with inputLevel adjustment
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}
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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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for (int i=0; i < 16; i++) {
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// fftResult[i] = (int)fftCalc[i];
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fftResult[i] = constrain((int)fftCalc[i],0,254); // question: why do we constrain values to 8bit here ???
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fftAvg[i] = (float)fftResult[i]*.05 + (1-.05)*fftAvg[i];
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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 * inputLevel/128 + (float)fftCalc[i]/16.0f; //with inputLevel adjustment
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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] = (int)fftCalc[i];
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fftResult[i] = constrain((int)fftCalc[i], 0, 254); // question: why do we constrain values to 8bit here ???
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fftAvg[i] = (float)fftResult[i]*0.05f + (1.0f - 0.05f)*fftAvg[i]; // why no just 0.95f*fftAvg[i]?
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}
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// release second sample to volume reactive effects.
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@ -364,7 +357,7 @@ class AudioReactive : public Usermod {
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int8_t i2sckPin = I2S_CKPIN;
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#endif
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WiFiUDP fftUdp;
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#define UDP_SYNC_HEADER "00001"
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struct audioSyncPacket {
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char header[6] = UDP_SYNC_HEADER;
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uint8_t myVals[32]; // 32 Bytes
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@ -377,8 +370,9 @@ class AudioReactive : public Usermod {
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double FFT_MajorPeak; // 08 Bytes
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};
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WiFiUDP fftUdp;
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// set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer)
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#define UDP_SYNC_HEADER "00001"
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uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger
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uint8_t binNum; // Used to select the bin for FFT based beat detection.
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@ -402,15 +396,16 @@ class AudioReactive : public Usermod {
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float beat = 0.f; // beat Detection
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float expAdjF; // Used for exponential filter.
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float weighting = 0.2; // Exponential filter weighting. Will be adjustable in a future release.
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float weighting = 0.2f; // Exponential filter weighting. Will be adjustable in a future release.
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bool udpSyncConnected = false;
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// strings to reduce flash memory usage (used more than twice)
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static const char _name[];
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static const char _analogmic[];
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static const char _digitalmic[];
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double mapf(double x, double in_min, double in_max, double out_min, double out_max);
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bool isValidUdpSyncVersion(char header[6]) {
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if (strncmp(header, UDP_SYNC_HEADER, 6) == 0) {
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return true;
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@ -419,6 +414,7 @@ class AudioReactive : public Usermod {
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}
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}
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void logAudio() {
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#ifdef MIC_LOGGER
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//Serial.print("micData:"); Serial.print(micData); Serial.print("\t");
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@ -500,6 +496,7 @@ class AudioReactive : public Usermod {
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#endif // FFT_SAMPLING_LOG
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} // logAudio()
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/*
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* A "PI control" multiplier to automatically adjust sound sensitivity.
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*
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@ -520,10 +517,10 @@ class AudioReactive : public Usermod {
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float tmpAgc = sampleReal * multAgc; // what-if amplified signal
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float control_error; // "control error" input for PI control
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static double control_integrated = 0.0; // "integrator control" = accumulated error
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static float control_integrated = 0.0f; // "integrator control" = accumulated error
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if (last_soundAgc != soundAgc)
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control_integrated = 0.0; // new preset - reset integrator
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control_integrated = 0.0f; // new preset - reset integrator
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// For PI control, we need to have a contant "frequency"
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// so let's make sure that the control loop is not running at insane speed
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@ -532,13 +529,13 @@ class AudioReactive : public Usermod {
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if (time_now - last_time > 2) {
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last_time = time_now;
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if((fabs(sampleReal) < 2.0) || (sampleMax < 1.0)) {
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if((fabs(sampleReal) < 2.0f) || (sampleMax < 1.0f)) {
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// MIC signal is "squelched" - deliver silence
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multAgcTemp = multAgc; // keep old control value (no change)
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tmpAgc = 0;
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// we need to "spin down" the intgrated error buffer
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if (fabs(control_integrated) < 0.01) control_integrated = 0.0;
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else control_integrated = control_integrated * 0.91;
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if (fabs(control_integrated) < 0.01f) control_integrated = 0.0f;
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else control_integrated = control_integrated * 0.91f;
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} else {
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// compute new setpoint
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if (tmpAgc <= agcTarget0Up[AGC_preset])
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@ -547,17 +544,17 @@ class AudioReactive : public Usermod {
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multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint
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}
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// limit amplification
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if (multAgcTemp > 32.0) multAgcTemp = 32.0;
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if (multAgcTemp < 1.0/64.0) multAgcTemp = 1.0/64.0;
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if (multAgcTemp > 32.0f) multAgcTemp = 32.0f;
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if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f;
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// compute error terms
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control_error = multAgcTemp - lastMultAgc;
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if (((multAgcTemp > 0.085) && (multAgcTemp < 6.5)) //integrator anti-windup by clamping
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if (((multAgcTemp > 0.085f) && (multAgcTemp < 6.5f)) //integrator anti-windup by clamping
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&& (multAgc*sampleMax < agcZoneStop[AGC_preset])) //integrator ceiling (>140% of max)
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control_integrated += control_error * 0.002 * 0.25; // 2ms = intgration time; 0.25 for damping
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control_integrated += control_error * 0.002f * 0.25f; // 2ms = intgration time; 0.25 for damping
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else
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control_integrated *= 0.9; // spin down that beasty integrator
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control_integrated *= 0.9f; // spin down that beasty integrator
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// apply PI Control
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tmpAgc = sampleReal * lastMultAgc; // check "zone" of the signal using previous gain
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@ -570,22 +567,22 @@ class AudioReactive : public Usermod {
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}
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// limit amplification again - PI controler sometimes "overshoots"
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if (multAgcTemp > 32.0) multAgcTemp = 32.0;
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if (multAgcTemp < 1.0/64.0) multAgcTemp = 1.0/64.0;
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if (multAgcTemp > 32.0f) multAgcTemp = 32.0f;
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if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f;
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}
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// NOW finally amplify the signal
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tmpAgc = sampleReal * multAgcTemp; // apply gain to signal
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if(fabs(sampleReal) < 2.0) tmpAgc = 0; // apply squelch threshold
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if(fabs(sampleReal) < 2.0f) tmpAgc = 0; // apply squelch threshold
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if (tmpAgc > 255) tmpAgc = 255; // limit to 8bit
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if (tmpAgc < 1) tmpAgc = 0; // just to be sure
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// update global vars ONCE - multAgc, sampleAGC, rawSampleAgc
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multAgc = multAgcTemp;
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rawSampleAgc = 0.8 * tmpAgc + 0.2 * (float)rawSampleAgc;
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rawSampleAgc = 0.8f * tmpAgc + 0.2f * (float)rawSampleAgc;
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// update smoothed AGC sample
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if(fabs(tmpAgc) < 1.0)
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sampleAgc = 0.5 * tmpAgc + 0.5 * sampleAgc; // fast path to zero
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if(fabs(tmpAgc) < 1.0f)
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sampleAgc = 0.5f * tmpAgc + 0.5f * sampleAgc; // fast path to zero
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else
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sampleAgc = sampleAgc + agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
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@ -595,6 +592,7 @@ class AudioReactive : public Usermod {
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last_soundAgc = soundAgc;
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} // agcAvg()
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void getSample() {
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static long peakTime;
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@ -617,8 +615,8 @@ class AudioReactive : public Usermod {
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#endif
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// Note to self: the next line kills 80% of sample - "miclev" filter runs at "full arduino loop" speed, following the signal almost instantly!
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//micLev = ((micLev * 31) + micIn) / 32; // Smooth it out over the last 32 samples for automatic centering
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micLev = ((micLev * 8191.0) + micDataReal) / 8192.0; // takes a few seconds to "catch up" with the Mic Input
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if(micIn < micLev) micLev = ((micLev * 31.0) + micDataReal) / 32.0; // align MicLev to lowest input signal
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micLev = ((micLev * 8191.0f) + micDataReal) / 8192.0f; // takes a few seconds to "catch up" with the Mic Input
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if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal
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micIn -= micLev; // Let's center it to 0 now
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/*---------DEBUG---------*/
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@ -646,17 +644,17 @@ class AudioReactive : public Usermod {
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sample = (int)sampleAdj; // ONLY update sample ONCE!!!!
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// keep "peak" sample, but decay value if current sample is below peak
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if ((sampleMax < sampleReal) && (sampleReal > 0.5)) {
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sampleMax = sampleMax + 0.5 * (sampleReal - sampleMax); // new peak - with some filtering
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if ((sampleMax < sampleReal) && (sampleReal > 0.5f)) {
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sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // new peak - with some filtering
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} else {
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if ((multAgc*sampleMax > agcZoneStop[AGC_preset]) && (soundAgc > 0))
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sampleMax = sampleMax + 0.5 * (sampleReal - sampleMax); // over AGC Zone - get back quickly
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else
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sampleMax = sampleMax * agcSampleDecay[AGC_preset]; // signal to zero --> 5-8sec
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if ((multAgc*sampleMax > agcZoneStop[AGC_preset]) && (soundAgc > 0))
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sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // over AGC Zone - get back quickly
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else
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sampleMax = sampleMax * agcSampleDecay[AGC_preset]; // signal to zero --> 5-8sec
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}
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if (sampleMax < 0.5) sampleMax = 0.0;
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if (sampleMax < 0.5f) sampleMax = 0.0f;
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sampleAvg = ((sampleAvg * 15.0) + sampleAdj) / 16.0; // Smooth it out over the last 16 samples.
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sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
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/*---------DEBUG---------*/
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DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(sample);
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@ -669,7 +667,7 @@ class AudioReactive : public Usermod {
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if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
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samplePeak = 0;
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udpSamplePeak = 0;
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}
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}
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if (userVar1 == 0) samplePeak = 0;
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// Poor man's beat detection by seeing if sample > Average + some value.
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@ -685,6 +683,7 @@ class AudioReactive : public Usermod {
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}
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} // getSample()
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/*
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* A simple averaging multiplier to automatically adjust sound sensitivity.
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*/
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@ -696,17 +695,17 @@ class AudioReactive : public Usermod {
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float lastMultAgc = multAgc;
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float tmpAgc;
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if(fabs(sampleAvg) < 2.0) {
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if(fabs(sampleAvg) < 2.0f) {
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tmpAgc = sampleAvg; // signal below squelch -> deliver silence
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multAgc = multAgc * 0.95; // slightly decrease gain multiplier
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multAgc = multAgc * 0.95f; // slightly decrease gain multiplier
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} else {
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multAgc = (sampleAvg < 1) ? targetAgc : targetAgc / sampleAvg; // Make the multiplier so that sampleAvg * multiplier = setpoint
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}
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if (multAgc < 0.5) multAgc = 0.5; // signal higher than 2*setpoint -> don't reduce it further
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multAgc = (lastMultAgc*127.0 +multAgc) / 128.0; //apply some filtering to gain multiplier -> smoother transitions
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if (multAgc < 0.5f) multAgc = 0.5f; // signal higher than 2*setpoint -> don't reduce it further
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multAgc = (lastMultAgc*127.0f +multAgc) / 128.0f; //apply some filtering to gain multiplier -> smoother transitions
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tmpAgc = (float)sample * multAgc; // apply gain to signal
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if (tmpAgc <= (soundSquelch*1.2)) tmpAgc = sample; // check against squelch threshold - increased by 20% to avoid artefacts (ripples)
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if (tmpAgc <= (soundSquelch*1.2f)) tmpAgc = sample; // check against squelch threshold - increased by 20% to avoid artefacts (ripples)
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if (tmpAgc > 255) tmpAgc = 255;
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sampleAgc = tmpAgc; // ONLY update sampleAgc ONCE because it's used elsewhere asynchronously!!!!
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@ -714,6 +713,7 @@ class AudioReactive : public Usermod {
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if (userVar0 > 255) userVar0 = 255;
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} // agcAvg()
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void transmitAudioData() {
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if (!udpSyncConnected) return;
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@ -852,7 +852,13 @@ class AudioReactive : public Usermod {
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* Use it to initialize network interfaces
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*/
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void connected() {
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//Serial.println("Connected to WiFi!");
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if (audioSyncPort > 0 || (audioSyncEnabled & 0x03)) {
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#ifndef ESP8266
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udpSyncConnected = fftUdp.beginMulticast(IPAddress(239, 0, 0, 1), audioSyncPort);
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#else
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udpSyncConnected = fftUdp.beginMulticast(WiFi.localIP(), IPAddress(239, 0, 0, 1), audioSyncPort);
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#endif
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}
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}
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@ -871,8 +877,7 @@ class AudioReactive : public Usermod {
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lastTime = millis();
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}
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if (!(audioSyncEnabled & (1 << 1))) { // Only run the sampling code IF we're not in Receive mode
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lastTime = millis();
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if (!(audioSyncEnabled & 0x02)) { // Only run the sampling code IF we're not in Receive mode
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if (soundAgc > AGC_NUM_PRESETS) soundAgc = 0; // make sure that AGC preset is valid (to avoid array bounds violation)
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getSample(); // Sample the microphone
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agcAvg(); // Calculated the PI adjusted value as sampleAvg
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@ -917,8 +922,8 @@ class AudioReactive : public Usermod {
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last_kick_time = now_time;
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}
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int new_user_inputLevel = 128.0 * multAgc; // scale AGC multiplier so that "1" is at 128
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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
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int new_user_inputLevel = 128.0f * multAgc; // scale AGC multiplier so that "1" is at 128
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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
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new_user_inputLevel = MIN(MAX(new_user_inputLevel, 0),255);
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// update user interfaces - restrict frequency to avoid flooding UI's with small changes
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@ -934,24 +939,22 @@ class AudioReactive : public Usermod {
|
||||
}
|
||||
lastMode = knownMode;
|
||||
|
||||
#if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
|
||||
#if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
|
||||
EVERY_N_MILLIS(20) {
|
||||
logAudio();
|
||||
}
|
||||
#endif
|
||||
|
||||
#endif
|
||||
}
|
||||
if (audioSyncEnabled & (1 << 0)) { // Only run the transmit code IF we're in Transmit mode
|
||||
//Serial.println("Transmitting UDP Mic Packet");
|
||||
|
||||
EVERY_N_MILLIS(20) {
|
||||
transmitAudioData();
|
||||
}
|
||||
|
||||
if (audioSyncEnabled & 0x01) { // Only run the transmit code IF we're in Transmit mode
|
||||
DEBUGSR_PRINTLN("Transmitting UDP Mic Packet");
|
||||
EVERY_N_MILLIS(20) {
|
||||
transmitAudioData();
|
||||
}
|
||||
}
|
||||
|
||||
// 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 (udpSyncConnected) {
|
||||
//Serial.println("Checking for UDP Microphone Packet");
|
||||
@ -1003,7 +1006,6 @@ class AudioReactive : public Usermod {
|
||||
}
|
||||
|
||||
|
||||
|
||||
bool getUMData(um_data_t **data) {
|
||||
if (!data) return false; // no pointer provided by caller -> exit
|
||||
*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).
|
||||
* 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.
|
||||
* 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)
|
||||
{
|
||||
JsonObject top = root.createNestedObject(FPSTR(_name));
|
||||
JsonArray pinArray = top.createNestedArray("pin");
|
||||
pinArray.add(audioPin);
|
||||
top[F("audio_pin")] = audioPin;
|
||||
//...
|
||||
|
||||
JsonObject amic = top.createNestedObject(FPSTR(_analogmic));
|
||||
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)
|
||||
{
|
||||
// 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)];
|
||||
|
||||
bool configComplete = !top.isNull();
|
||||
|
||||
configComplete &= getJsonValue(top[F("audio_pin")], audioPin);
|
||||
/*
|
||||
configComplete &= getJsonValue(top["testBool"], testBool);
|
||||
configComplete &= getJsonValue(top["testULong"], testULong);
|
||||
configComplete &= getJsonValue(top["testFloat"], testFloat);
|
||||
configComplete &= getJsonValue(top["testString"], testString);
|
||||
configComplete &= getJsonValue(top[FPSTR(_analogmic)]["pin"], audioPin);
|
||||
|
||||
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["type"], dmType);
|
||||
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][0], i2ssdPin);
|
||||
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][1], i2swsPin);
|
||||
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;
|
||||
}
|
||||
|
||||
@ -1163,3 +1186,5 @@ class AudioReactive : public Usermod {
|
||||
|
||||
// strings to reduce flash memory usage (used more than twice)
|
||||
const char AudioReactive::_name[] PROGMEM = "AudioReactive";
|
||||
const char AudioReactive::_analogmic[] PROGMEM = "analogmic";
|
||||
const char AudioReactive::_digitalmic[] PROGMEM = "digitalmic";
|
||||
|
||||
Loading…
Reference in New Issue
Block a user