Merge branch 'mdev' into ES8388
This commit is contained in:
Will Tatam
@@ -29,9 +29,9 @@
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// #define SR_DEBUG // generic SR DEBUG messages
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#ifdef SR_DEBUG
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#define DEBUGSR_PRINT(x) DEBUGOUT.print(x)
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#define DEBUGSR_PRINTLN(x) DEBUGOUT.println(x)
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#define DEBUGSR_PRINTF(x...) DEBUGOUT.printf(x)
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#define DEBUGSR_PRINT(x) DEBUGOUT(x)
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#define DEBUGSR_PRINTLN(x) DEBUGOUTLN(x)
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#define DEBUGSR_PRINTF(x...) DEBUGOUTF(x)
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#else
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#define DEBUGSR_PRINT(x)
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#define DEBUGSR_PRINTLN(x)
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@@ -55,10 +55,10 @@
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#endif
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#if defined(MIC_LOGGER) || defined(FFT_SAMPLING_LOG)
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#define PLOT_PRINT(x) DEBUGOUT.print(x)
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#define PLOT_PRINTLN(x) DEBUGOUT.println(x)
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#define PLOT_PRINTF(x...) DEBUGOUT.printf(x)
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#define PLOT_FLUSH() DEBUGOUT.flush()
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#define PLOT_PRINT(x) DEBUGOUT(x)
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#define PLOT_PRINTLN(x) DEBUGOUTLN(x)
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#define PLOT_PRINTF(x...) DEBUGOUTF(x)
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#define PLOT_FLUSH() DEBUGOUTFlush()
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#else
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#define PLOT_PRINT(x)
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#define PLOT_PRINTLN(x)
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@@ -123,6 +123,15 @@ static AudioSource *audioSource = nullptr;
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static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks.
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static bool useBandPassFilter = false; // if true, enables a bandpass filter 80Hz-16Khz to remove noise. Applies before FFT.
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//WLEDMM add experimental settings
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static uint8_t micLevelMethod = 0; // 0=old "floating" miclev, 1=new "freeze" mode, 2=fast freeze mode (mode 2 may not work for you)
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#if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3)
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static uint8_t averageByRMS = false; // false: use mean value, true: use RMS (root mean squared). use simpler method on slower MCUs.
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#else
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static uint8_t averageByRMS = true; // false: use mean value, true: use RMS (root mean squared). use better method on fast MCUs.
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#endif
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static uint8_t freqDist = 0; // 0=old 1=rightshift mode
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// audioreactive variables shared with FFT task
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static float micDataReal = 0.0f; // MicIn data with full 24bit resolution - lowest 8bit after decimal point
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static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier
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@@ -136,7 +145,7 @@ static float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending
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// peak detection
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static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay()
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static uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
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static uint8_t maxVol = 31; // (was 10) Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
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static uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated)
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static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
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static unsigned long timeOfPeak = 0; // time of last sample peak detection.
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@@ -182,7 +191,7 @@ static const float fftResultPink[MAX_PINK+1][NUM_GEQ_CHANNELS] = {
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{ 12.0f, 6.60f, 2.60f, 1.15f, 1.35f, 2.05f, 2.85f, 2.50f, 2.85f, 3.30f, 2.25f, 4.35f, 3.80f, 3.75f, 6.50f, 9.00f}, // 4 IMNP441 - voice, or small speaker
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{ 2.75f, 1.60f, 1.40f, 1.46f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 1.75f, 2.55f, 3.60f }, // 5 ICS-43434 datasheet response * pink noise
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{ 2.25f, 1.20f, 1.00f, 1.20f, 1.80f, 3.20f, 5.10f, 5.50f, 4.00f, 4.80f, 6.70f, 6.40f, 5.80f, 3.90f, 6.00f, 5.10f }, // 6 ICS-43434 - big speaker, strong bass
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{ 2.90f, 1.25f, 0.75f, 1.08f, 2.35f, 3.55f, 3.60f, 3.40f, 2.75f, 3.45f, 4.40f, 6.35f, 6.80f, 6.80f, 8.50f,10.64f }, // 6 ICS-43434 - big speaker, strong bass
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{ 1.65f, 1.00f, 1.05f, 1.30f, 1.48f, 1.30f, 1.80f, 3.00f, 1.50f, 1.65f, 2.56f, 3.00f, 2.60f, 2.30f, 5.00f, 3.00f }, // 7 SPM1423
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{ 2.25f, 1.60f, 1.30f, 1.60f, 2.20f, 3.20f, 3.06f, 2.60f, 2.85f, 3.50f, 4.10f, 4.80f, 5.70f, 6.05f,10.50f,14.85f }, // 8 userdef #1 for ewowi (enhance median/high freqs)
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@@ -264,7 +273,8 @@ constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - Thi
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constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information.
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// the following are observed values, supported by a bit of "educated guessing"
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//#define FFT_DOWNSCALE 0.65f // 20kHz - downscaling factor for FFT results - "Flat-Top" window @20Khz, old freq channels
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#define FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels
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//#define FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels
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#define FFT_DOWNSCALE 0.40f // downscaling factor for FFT results, RMS averaging
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#define LOG_256 5.54517744f // log(256)
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// These are the input and output vectors. Input vectors receive computed results from FFT.
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@@ -303,23 +313,28 @@ static float mapf(float x, float in_min, float in_max, float out_min, float out_
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}
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// compute average of several FFT resut bins
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#if 1 // linear average
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static float fftAddAvg(int from, int to) {
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// linear average
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static float fftAddAvgLin(int from, int to) {
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float result = 0.0f;
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for (int i = from; i <= to; i++) {
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result += vReal[i];
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}
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return result / float(to - from + 1);
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}
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#else // RMS average
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static float fftAddAvg(int from, int to) {
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// RMS average
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static float fftAddAvgRMS(int from, int to) {
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double result = 0.0;
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for (int i = from; i <= to; i++) {
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result += vReal[i] * vReal[i];
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}
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return sqrtf(result / float(to - from + 1));
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}
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#endif
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static float fftAddAvg(int from, int to) {
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if (from == to) return vReal[from]; // small optimization
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if (averageByRMS) return fftAddAvgRMS(from, to); // use SMS
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else return fftAddAvgLin(from, to); // use linear average
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}
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#if defined(CONFIG_IDF_TARGET_ESP32C3)
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constexpr bool skipSecondFFT = true;
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@@ -512,35 +527,67 @@ void FFTcode(void * parameter)
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fftCalc[14] = fftAddAvg(147,194); // 2940 - 3900
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fftCalc[15] = fftAddAvg(194,250); // 3880 - 5000 // avoid the last 5 bins, which are usually inaccurate
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#else
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/* new mapping, optimized for 22050 Hz by softhack007 */
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//WLEDMM: different distributions
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if (freqDist == 0) {
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/* new mapping, optimized for 22050 Hz by softhack007 --- update: removed overlap */
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// bins frequency range
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if (useBandPassFilter) {
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// skip frequencies below 100hz
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fftCalc[ 0] = 0.8f * fftAddAvg(3,4);
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fftCalc[ 1] = 0.9f * fftAddAvg(4,5);
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fftCalc[ 2] = fftAddAvg(5,6);
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fftCalc[ 3] = fftAddAvg(6,7);
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fftCalc[ 0] = 0.8f * fftAddAvg(3,3);
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fftCalc[ 1] = 0.9f * fftAddAvg(4,4);
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fftCalc[ 2] = fftAddAvg(5,5);
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fftCalc[ 3] = fftAddAvg(6,6);
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// don't use the last bins from 206 to 255.
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fftCalc[15] = fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping
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} else {
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fftCalc[ 0] = fftAddAvg(1,2); // 1 43 - 86 sub-bass
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fftCalc[ 1] = fftAddAvg(2,3); // 1 86 - 129 bass
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fftCalc[ 2] = fftAddAvg(3,5); // 2 129 - 216 bass
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fftCalc[ 3] = fftAddAvg(5,7); // 2 216 - 301 bass + midrange
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fftCalc[ 0] = fftAddAvg(1,1); // 1 43 - 86 sub-bass
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fftCalc[ 1] = fftAddAvg(2,2); // 1 86 - 129 bass
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fftCalc[ 2] = fftAddAvg(3,4); // 2 129 - 216 bass
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fftCalc[ 3] = fftAddAvg(5,6); // 2 216 - 301 bass + midrange
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// don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise)
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fftCalc[15] = fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
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}
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fftCalc[ 4] = fftAddAvg(7,10); // 3 301 - 430 midrange
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fftCalc[ 5] = fftAddAvg(10,13); // 3 430 - 560 midrange
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fftCalc[ 6] = fftAddAvg(13,19); // 5 560 - 818 midrange
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fftCalc[ 7] = fftAddAvg(19,26); // 7 818 - 1120 midrange -- 1Khz should always be the center !
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fftCalc[ 8] = fftAddAvg(26,33); // 7 1120 - 1421 midrange
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fftCalc[ 9] = fftAddAvg(33,44); // 9 1421 - 1895 midrange
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fftCalc[10] = fftAddAvg(44,56); // 12 1895 - 2412 midrange + high mid
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fftCalc[11] = fftAddAvg(56,70); // 14 2412 - 3015 high mid
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fftCalc[12] = fftAddAvg(70,86); // 16 3015 - 3704 high mid
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fftCalc[13] = fftAddAvg(86,104); // 18 3704 - 4479 high mid
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fftCalc[14] = fftAddAvg(104,165) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
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fftCalc[ 4] = fftAddAvg(7,9); // 3 301 - 430 midrange
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fftCalc[ 5] = fftAddAvg(10,12); // 3 430 - 560 midrange
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fftCalc[ 6] = fftAddAvg(13,18); // 5 560 - 818 midrange
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fftCalc[ 7] = fftAddAvg(19,25); // 7 818 - 1120 midrange -- 1Khz should always be the center !
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fftCalc[ 8] = fftAddAvg(26,32); // 7 1120 - 1421 midrange
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fftCalc[ 9] = fftAddAvg(33,43); // 9 1421 - 1895 midrange
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fftCalc[10] = fftAddAvg(44,55); // 12 1895 - 2412 midrange + high mid
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fftCalc[11] = fftAddAvg(56,69); // 14 2412 - 3015 high mid
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fftCalc[12] = fftAddAvg(70,85); // 16 3015 - 3704 high mid
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fftCalc[13] = fftAddAvg(86,103); // 18 3704 - 4479 high mid
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fftCalc[14] = fftAddAvg(104,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
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}
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else if (freqDist == 1) { //WLEDMM: Rightshft: note ewowi: frequencies in comments are not correct
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if (useBandPassFilter) {
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// skip frequencies below 100hz
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fftCalc[ 0] = 0.8f * fftAddAvg(1,1);
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fftCalc[ 1] = 0.9f * fftAddAvg(2,2);
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fftCalc[ 2] = fftAddAvg(3,3);
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fftCalc[ 3] = fftAddAvg(4,4);
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// don't use the last bins from 206 to 255.
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fftCalc[15] = fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping
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} else {
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fftCalc[ 0] = fftAddAvg(1,1); // 1 43 - 86 sub-bass
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fftCalc[ 1] = fftAddAvg(2,2); // 1 86 - 129 bass
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fftCalc[ 2] = fftAddAvg(3,3); // 2 129 - 216 bass
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fftCalc[ 3] = fftAddAvg(4,4); // 2 216 - 301 bass + midrange
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// don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise)
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fftCalc[15] = fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
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}
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fftCalc[ 4] = fftAddAvg(5,6); // 3 301 - 430 midrange
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fftCalc[ 5] = fftAddAvg(7,8); // 3 430 - 560 midrange
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fftCalc[ 6] = fftAddAvg(9,10); // 5 560 - 818 midrange
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fftCalc[ 7] = fftAddAvg(11,13); // 7 818 - 1120 midrange -- 1Khz should always be the center !
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fftCalc[ 8] = fftAddAvg(14,18); // 7 1120 - 1421 midrange
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fftCalc[ 9] = fftAddAvg(19,25); // 9 1421 - 1895 midrange
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fftCalc[10] = fftAddAvg(26,36); // 12 1895 - 2412 midrange + high mid
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fftCalc[11] = fftAddAvg(37,45); // 14 2412 - 3015 high mid
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fftCalc[12] = fftAddAvg(46,66); // 16 3015 - 3704 high mid
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fftCalc[13] = fftAddAvg(67,97); // 18 3704 - 4479 high mid
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fftCalc[14] = fftAddAvg(98,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
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}
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#endif
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} else { // noise gate closed - just decay old values
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isFirstRun = false;
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@@ -861,7 +908,9 @@ class AudioReactive : public Usermod {
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static const char _name[];
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static const char _enabled[];
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static const char _inputLvl[];
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#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
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static const char _analogmic[];
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#endif
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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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@@ -1057,6 +1106,10 @@ class AudioReactive : public Usermod {
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const float weighting = 0.2f; // Exponential filter weighting. Will be adjustable in a future release.
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const float weighting2 = 0.073f; // Exponential filter weighting, for rising signal (a bit more robust against spikes)
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const int AGC_preset = (soundAgc > 0)? (soundAgc-1): 0; // make sure the _compiler_ knows this value will not change while we are inside the function
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static bool isFrozen = false;
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static bool haveSilence = true;
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static unsigned long lastSoundTime = 0; // for delaying un-freeze
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static unsigned long startuptime = 0; // "fast freeze" mode: do not interfere during first 12 seconds (filter startup time)
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#ifdef WLED_DISABLE_SOUND
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micIn = inoise8(millis(), millis()); // Simulated analog read
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@@ -1079,8 +1132,15 @@ class AudioReactive : public Usermod {
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#endif
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#endif
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micLev += (micDataReal-micLev) / 12288.0f;
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if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal
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if (startuptime == 0) startuptime = millis(); // fast freeze mode - remember filter startup time
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if ((micLevelMethod < 1) || !isFrozen) { // following the input level, UNLESS mic Level was frozen
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micLev += (micDataReal-micLev) / 12288.0f;
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}
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if(micDataReal < (micLev-0.24)) { // MicLev above input signal:
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micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // always align MicLev to lowest input signal
|
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if (!haveSilence) isFrozen = true; // freeze mode: freeze micLevel so it cannot rise again
|
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}
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micIn -= micLev; // Let's center it to 0 now
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// Using an exponential filter to smooth out the signal. We'll add controls for this in a future release.
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@@ -1093,10 +1153,26 @@ class AudioReactive : public Usermod {
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expAdjF = fabsf(expAdjF); // Now (!) take the absolute value
|
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|
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if ((micLevelMethod == 2) && !haveSilence && (expAdjF >= (1.5f * float(soundSquelch))))
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isFrozen = true; // fast freeze mode: freeze micLevel once the volume rises 50% above squelch
|
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//expAdjF = (micInNoDC <= soundSquelch) ? 0: expAdjF; // simple noise gate - experimental
|
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expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate
|
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if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0"
|
||||
|
||||
if (expAdjF <= 0.5f)
|
||||
haveSilence = true;
|
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else {
|
||||
lastSoundTime = millis();
|
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haveSilence = false;
|
||||
}
|
||||
|
||||
// un-freeze micLev
|
||||
if (micLevelMethod == 0) isFrozen = false;
|
||||
if ((micLevelMethod == 1) && isFrozen && haveSilence && ((millis() - lastSoundTime) > 4000)) isFrozen = false; // normal freeze: 4 seconds silence needed
|
||||
if ((micLevelMethod == 2) && isFrozen && haveSilence && ((millis() - lastSoundTime) > 6000)) isFrozen = false; // fast freeze: 6 seconds silence needed
|
||||
if ((micLevelMethod == 2) && (millis() - startuptime < 12000)) isFrozen = false; // fast freeze: no freeze in first 12 seconds (filter startup phase)
|
||||
|
||||
tmpSample = expAdjF;
|
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micIn = abs(micIn); // And get the absolute value of each sample
|
||||
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@@ -1360,6 +1436,7 @@ class AudioReactive : public Usermod {
|
||||
case 0: //ADC analog
|
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#if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3)
|
||||
case 5: //PDM Microphone
|
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case 51: //legacy PDM Microphone
|
||||
#endif
|
||||
#endif
|
||||
case 1:
|
||||
@@ -1397,6 +1474,13 @@ class AudioReactive : public Usermod {
|
||||
delay(100);
|
||||
if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin);
|
||||
break;
|
||||
case 51:
|
||||
DEBUGSR_PRINT(F("AR: Legacy PDM Microphone - ")); DEBUGSR_PRINTLN(F(I2S_PDM_MIC_CHANNEL_TEXT));
|
||||
audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 1.0f);
|
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useBandPassFilter = true;
|
||||
delay(100);
|
||||
if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin);
|
||||
break;
|
||||
#endif
|
||||
case 6:
|
||||
DEBUGSR_PRINTLN(F("AR: ES8388 Source"));
|
||||
@@ -1537,7 +1621,7 @@ class AudioReactive : public Usermod {
|
||||
|
||||
#if defined(WLED_DEBUG) || defined(SR_DEBUG) || defined(SR_STATS)
|
||||
// complain when audio userloop has been delayed for long time. Currently we need userloop running between 500 and 1500 times per second.
|
||||
if ((userloopDelay > /*23*/ 30) && !disableSoundProcessing && (audioSyncEnabled == 0)) {
|
||||
if ((userloopDelay > /*23*/ 65) && !disableSoundProcessing && (audioSyncEnabled == 0)) {
|
||||
USER_PRINTF("[AR userLoop] hickup detected -> was inactive for last %d millis!\n", userloopDelay);
|
||||
}
|
||||
#endif
|
||||
@@ -1673,14 +1757,15 @@ class AudioReactive : public Usermod {
|
||||
connected(); // resume UDP
|
||||
} else
|
||||
// xTaskCreatePinnedToCore(
|
||||
xTaskCreate( // no need to "pin" this task to core #0
|
||||
// xTaskCreate( // no need to "pin" this task to core #0
|
||||
xTaskCreateUniversal(
|
||||
FFTcode, // Function to implement the task
|
||||
"FFT", // Name of the task
|
||||
5000, // Stack size in words
|
||||
NULL, // Task input parameter
|
||||
1, // Priority of the task
|
||||
&FFT_Task // Task handle
|
||||
// , 0 // Core where the task should run
|
||||
, 0 // Core where the task should run
|
||||
);
|
||||
}
|
||||
micDataReal = 0.0f; // just to be sure
|
||||
@@ -1780,7 +1865,10 @@ class AudioReactive : public Usermod {
|
||||
if (audioSource->getType() == AudioSource::Type_I2SAdc) {
|
||||
infoArr.add(F("ADC analog"));
|
||||
} else {
|
||||
infoArr.add(F("I2S digital"));
|
||||
if (dmType != 51)
|
||||
infoArr.add(F("I2S digital"));
|
||||
else
|
||||
infoArr.add(F("legacy I2S PDM"));
|
||||
}
|
||||
// input level or "silence"
|
||||
if (maxSample5sec > 1.0) {
|
||||
@@ -1793,7 +1881,7 @@ class AudioReactive : public Usermod {
|
||||
} else {
|
||||
// error during audio source setup
|
||||
infoArr.add(F("not initialized"));
|
||||
infoArr.add(F(" - check GPIO config"));
|
||||
infoArr.add(F(" - check pin settings"));
|
||||
}
|
||||
}
|
||||
|
||||
@@ -1957,6 +2045,12 @@ class AudioReactive : public Usermod {
|
||||
cfg[F("gain")] = sampleGain;
|
||||
cfg[F("AGC")] = soundAgc;
|
||||
|
||||
//WLEDMM: experimental settings
|
||||
JsonObject poweruser = top.createNestedObject("experiments");
|
||||
poweruser[F("micLev")] = micLevelMethod;
|
||||
poweruser[F("freqDist")] = freqDist;
|
||||
poweruser[F("freqRMS")] = averageByRMS;
|
||||
|
||||
JsonObject dynLim = top.createNestedObject("dynamics");
|
||||
dynLim[F("limiter")] = limiterOn;
|
||||
dynLim[F("rise")] = attackTime;
|
||||
@@ -2005,7 +2099,11 @@ class AudioReactive : public Usermod {
|
||||
if (dmType == 0) dmType = SR_DMTYPE; // MCU does not support analog
|
||||
#if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3)
|
||||
if (dmType == 5) dmType = SR_DMTYPE; // MCU does not support PDM
|
||||
if (dmType == 51) dmType = SR_DMTYPE; // MCU does not support legacy PDM
|
||||
#endif
|
||||
#else
|
||||
if (dmType == 5) useBandPassFilter = true; // enable filter for PDM
|
||||
if (dmType == 51) useBandPassFilter = true /*false*/; // switch on filter for legacy PDM
|
||||
#endif
|
||||
|
||||
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][0], i2ssdPin);
|
||||
@@ -2019,6 +2117,11 @@ class AudioReactive : public Usermod {
|
||||
configComplete &= getJsonValue(top["config"][F("gain")], sampleGain);
|
||||
configComplete &= getJsonValue(top["config"][F("AGC")], soundAgc);
|
||||
|
||||
//WLEDMM: experimental settings
|
||||
configComplete &= getJsonValue(top["experiments"][F("micLev")], micLevelMethod);
|
||||
configComplete &= getJsonValue(top["experiments"][F("freqDist")], freqDist);
|
||||
configComplete &= getJsonValue(top["experiments"][F("freqRMS")], averageByRMS);
|
||||
|
||||
configComplete &= getJsonValue(top["dynamics"][F("limiter")], limiterOn);
|
||||
configComplete &= getJsonValue(top["dynamics"][F("rise")], attackTime);
|
||||
configComplete &= getJsonValue(top["dynamics"][F("fall")], decayTime);
|
||||
@@ -2038,10 +2141,12 @@ class AudioReactive : public Usermod {
|
||||
oappend(SET_F("addInfo('AudioReactive:help',0,'<button onclick=\"location.href="https://mm.kno.wled.ge/soundreactive/Sound-Settings"\" type=\"button\">?</button>');"));
|
||||
|
||||
//WLEDMM: add defaults
|
||||
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) // -S3/-S2/-C3 don't support analog audio
|
||||
#ifdef AUDIOPIN
|
||||
oappend(SET_F("xOpt('AudioReactive:analogmic:pin',1,' ⎌',")); oappendi(AUDIOPIN); oappend(");");
|
||||
#endif
|
||||
oappend(SET_F("aOpt('AudioReactive:analogmic:pin',1);")); //only analog options
|
||||
#endif
|
||||
|
||||
oappend(SET_F("dd=addDropdown('AudioReactive','digitalmic:type');"));
|
||||
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
|
||||
@@ -2071,12 +2176,17 @@ class AudioReactive : public Usermod {
|
||||
#else
|
||||
oappend(SET_F("addOption(dd,'Generic I2S with Mclk',4);"));
|
||||
#endif
|
||||
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
|
||||
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
|
||||
#if SR_DMTYPE==5
|
||||
oappend(SET_F("addOption(dd,'Generic I2S PDM (⎌)',5);"));
|
||||
#else
|
||||
oappend(SET_F("addOption(dd,'Generic I2S PDM',5);"));
|
||||
#endif
|
||||
#if SR_DMTYPE==51
|
||||
oappend(SET_F("addOption(dd,'.Legacy I2S PDM ☾ (⎌)',51);"));
|
||||
#else
|
||||
oappend(SET_F("addOption(dd,'.Legacy I2S PDM ☾',51);"));
|
||||
#endif
|
||||
#endif
|
||||
oappend(SET_F("addOption(dd,'ES8388',6);"));
|
||||
|
||||
@@ -2093,6 +2203,23 @@ class AudioReactive : public Usermod {
|
||||
oappend(SET_F("addOption(dd,'Vivid',2);"));
|
||||
oappend(SET_F("addOption(dd,'Lazy',3);"));
|
||||
|
||||
//WLEDMM: experimental settings
|
||||
oappend(SET_F("dd=addDropdown('AudioReactive','experiments:micLev');"));
|
||||
oappend(SET_F("addOption(dd,'Floating (⎌)',0);"));
|
||||
oappend(SET_F("addOption(dd,'Freeze',1);"));
|
||||
oappend(SET_F("addOption(dd,'Fast Freeze',2);"));
|
||||
oappend(SET_F("addInfo('AudioReactive:experiments:micLev',1,'☾');"));
|
||||
|
||||
oappend(SET_F("dd=addDropdown('AudioReactive','experiments:freqDist');"));
|
||||
oappend(SET_F("addOption(dd,'Normal (⎌)',0);"));
|
||||
oappend(SET_F("addOption(dd,'RightShift',1);"));
|
||||
oappend(SET_F("addInfo('AudioReactive:experiments:freqDist',1,'☾');"));
|
||||
|
||||
oappend(SET_F("dd=addDropdown('AudioReactive','experiments:freqRMS');"));
|
||||
oappend(SET_F("addOption(dd,'Off (⎌)',0);"));
|
||||
oappend(SET_F("addOption(dd,'On',1);"));
|
||||
oappend(SET_F("addInfo('AudioReactive:experiments:freqRMS',1,'☾');"));
|
||||
|
||||
oappend(SET_F("dd=addDropdown('AudioReactive','dynamics:limiter');"));
|
||||
oappend(SET_F("addOption(dd,'Off',0);"));
|
||||
oappend(SET_F("addOption(dd,'On',1);"));
|
||||
@@ -2163,6 +2290,7 @@ class AudioReactive : public Usermod {
|
||||
#else
|
||||
oappend(SET_F("addOption(dd,'userdefined #2',9);"));
|
||||
#endif
|
||||
oappend(SET_F("addInfo('AudioReactive:frequency:profile',1,'☾');"));
|
||||
|
||||
oappend(SET_F("dd=addDropdown('AudioReactive','sync:mode');"));
|
||||
oappend(SET_F("addOption(dd,'Off',0);"));
|
||||
@@ -2237,7 +2365,9 @@ class AudioReactive : public Usermod {
|
||||
const char AudioReactive::_name[] PROGMEM = "AudioReactive";
|
||||
const char AudioReactive::_enabled[] PROGMEM = "enabled";
|
||||
const char AudioReactive::_inputLvl[] PROGMEM = "inputLevel";
|
||||
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
|
||||
const char AudioReactive::_analogmic[] PROGMEM = "analogmic";
|
||||
#endif
|
||||
const char AudioReactive::_digitalmic[] PROGMEM = "digitalmic";
|
||||
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
|
||||
|
||||
Reference in New Issue
Block a user