diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 40c17616..e5bda203 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -24,7 +24,7 @@ // #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter) // #define FFT_SAMPLING_LOG // FFT result debugging // #define SR_DEBUG // generic SR DEBUG messages -// #define NO_MIC_LOGGER // exclude MIC_LOGGER from SR_DEBUG + #ifdef SR_DEBUG #define DEBUGSR_PRINT(x) Serial.print(x) @@ -85,18 +85,25 @@ const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; // static AudioSource *audioSource = nullptr; static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks. +// audioreactive variables shared with FFT task static float micDataReal = 0.0f; // MicIn data with full 24bit resolution - lowest 8bit after decimal point -static float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC. static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier +static float sampleAvg = 0.0f; // Smoothed Average sample - sampleAvg < 1 means "quiet" (simple noise gate) + +// peak detection +static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay() +static uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated) +static uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated) +static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData +static unsigned long timeOfPeak = 0; // time of last sample peak detection. +static void detectSamplePeak(void); // peak detection function (needs scaled FFT reasults in vReal[]) +static void autoResetPeak(void); // peak auto-reset function -static int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel) -static int16_t rawSampleAgc = 0; // not smoothed AGC sample -static float sampleAvg = 0.0f; // Smoothed Average sampleRaw -static float sampleAgc = 0.0f; // Smoothed AGC sample //////////////////// // Begin FFT Code // //////////////////// + #ifdef UM_AUDIOREACTIVE_USE_NEW_FFT // lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2 #define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc), and an a few other speedups @@ -105,21 +112,22 @@ static float sampleAgc = 0.0f; // Smoothed AGC sample #endif #include "arduinoFFT.h" -// FFT Variables +// FFT Output variables shared with animations +#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !! +static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency +static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency +static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects + +// FFT Constants constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - This value MUST ALWAYS be a power of 2 constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information. -static float FFT_MajorPeak = 1.0f; -static float FFT_Magnitude = 0.0f; - // These are the input and output vectors. Input vectors receive computed results from FFT. -static float vReal[samplesFFT] = {0.0f}; -static float vImag[samplesFFT] = {0.0f}; -static float fftBin[samplesFFT_2] = {0.0f}; +static float vReal[samplesFFT] = {0.0f}; // FFT sample inputs / freq output - these are our raw result bins +static float vImag[samplesFFT] = {0.0f}; // imaginary parts // the following are observed values, supported by a bit of "educated guessing" //#define FFT_DOWNSCALE 0.65f // 20kHz - downscaling factor for FFT results - "Flat-Top" window @20Khz, old freq channels - #define FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels #define LOG_256 5.54517744 @@ -128,13 +136,11 @@ static float windowWeighingFactors[samplesFFT] = {0.0f}; #endif // Try and normalize fftBin values to a max of 4096, so that 4096/16 = 256. -// Oh, and bins 0,1,2 are no good, so we'll zero them out. -static float fftCalc[16] = {0.0f}; -static uint8_t fftResult[16] = {0}; // Our calculated result table, which we feed to the animations. +static float fftCalc[NUM_GEQ_CHANNELS] = {0.0f}; +static float fftAvg[NUM_GEQ_CHANNELS] = {0.0f}; // Calculated frequency channel results, with smoothing (used if dynamics limiter is ON) #ifdef SR_DEBUG -static float fftResultMax[16] = {0.0f}; // A table used for testing to determine how our post-processing is working. +static float fftResultMax[NUM_GEQ_CHANNELS] = {0.0f}; // A table used for testing to determine how our post-processing is working. #endif -static float fftAvg[16] = {0.0f}; #ifdef WLED_DEBUG static unsigned long fftTime = 0; @@ -142,7 +148,7 @@ static unsigned long sampleTime = 0; #endif // Table of multiplication factors so that we can even out the frequency response. -static float fftResultPink[16] = { 1.70f, 1.71f, 1.73f, 1.78f, 1.68f, 1.56f, 1.55f, 1.63f, 1.79f, 1.62f, 1.80f, 2.06f, 2.47f, 3.35f, 6.83f, 9.55f }; +static float fftResultPink[NUM_GEQ_CHANNELS] = { 1.70f, 1.71f, 1.73f, 1.78f, 1.68f, 1.56f, 1.55f, 1.63f, 1.79f, 1.62f, 1.80f, 2.06f, 2.47f, 3.35f, 6.83f, 9.55f }; // Create FFT object #ifdef UM_AUDIOREACTIVE_USE_NEW_FFT @@ -161,12 +167,12 @@ static float mapf(float x, float in_min, float in_max, float out_min, float out_ static float fftAddAvg(int from, int to) { float result = 0.0f; for (int i = from; i <= to; i++) { - result += fftBin[i]; + result += vReal[i]; } return result / float(to - from + 1); } -// FFT main code +// FFT main task void FFTcode(void * parameter) { DEBUGSR_PRINT("FFT started on core: "); DEBUGSR_PRINTLN(xPortGetCoreID()); @@ -237,9 +243,9 @@ void FFTcode(void * parameter) #endif FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects - for (int i = 0; i < samplesFFT_2; i++) { // Values for bins 0 and 1 are WAY too large. Might as well start at 3. + for (int i = 0; i < samplesFFT; i++) { float t = fabsf(vReal[i]); // just to be sure - values in fft bins should be positive any way - fftBin[i] = t / 16.0f; // Reduce magnitude. Want end result to be linear and ~4096 max. + vReal[i] = t / 16.0f; // Reduce magnitude. Want end result to be scaled linear and ~4096 max. } // for() // mapping of FFT result bins to frequency channels @@ -292,14 +298,14 @@ void FFTcode(void * parameter) // don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise) #endif } else { // noise gate closed - just decay old values - for (int i=0; i < 16; i++) { + for (int i=0; i < NUM_GEQ_CHANNELS; i++) { fftCalc[i] *= 0.85f; // decay to zero if (fftCalc[i] < 4.0f) fftCalc[i] = 0.0f; } } // post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling) - for (int i=0; i < 16; i++) { + for (int i=0; i < NUM_GEQ_CHANNELS; i++) { if (sampleAvg > 1) { // noise gate open // Adjustment for frequency curves. @@ -378,11 +384,43 @@ void FFTcode(void * parameter) fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth } #endif + // run peak detection + autoResetPeak(); + detectSamplePeak(); - } // for(;;) -} // FFTcode() + } // for(;;)ever +} // FFTcode() task end +//////////////////// +// Peak detection // +//////////////////// + +// peak detection is called from FFT task when vReal[] contains valid FFT results +static void detectSamplePeak(void) { + // Poor man's beat detection by seeing if sample > Average + some value. + if ((sampleAvg > 1) && (maxVol > 0) && (binNum > 1) && (vReal[binNum] > maxVol) && ((millis() - timeOfPeak) > 100)) { + // This goes through ALL of the 255 bins - but ignores stupid settings + // Then we got a peak, else we don't. The peak has to time out on its own in order to support UDP sound sync. + samplePeak = true; + timeOfPeak = millis(); + udpSamplePeak = true; + } +} + +static void autoResetPeak(void) { + uint16_t MinShowDelay = MAX(50, strip.getMinShowDelay()); // Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC + if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed. + samplePeak = false; + if (audioSyncEnabled == 0) udpSamplePeak = false; // this is normally reset by transmitAudioData + } +} + + +//////////////////// +// usermod class // +//////////////////// + //class name. Use something descriptive and leave the ": public Usermod" part :) class AudioReactive : public Usermod { @@ -453,40 +491,36 @@ class AudioReactive : public Usermod { 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) bool enabled = false; bool initDone = false; - const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED + // variables for UDP sound sync + WiFiUDP fftUdp; // UDP object for sound sync (from WiFi UDP, not Async UDP!) + bool udpSyncConnected = false;// UDP connection status -> true if connected to multicast group + unsigned long lastTime = 0; // last time of running UDP Microphone Sync + const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED + uint16_t audioSyncPort= 11988;// default port for UDP sound sync + + // used for AGC + int last_soundAgc = -1; // used to detect AGC mode change (for resetting AGC internal error buffers) + double control_integrated = 0.0; // persistent across calls to agcAvg(); "integrator control" = accumulated error + + // variables used by getSample() and agcAvg() + int16_t micIn = 0; // Current sample starts with negative values and large values, which is why it's 16 bit signed + double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controler. + float micLev = 0.0f; // Used to convert returned value to have '0' as minimum. A leveller + float expAdjF = 0.0f; // Used for exponential filter. + float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC. + int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel) + int16_t rawSampleAgc = 0; // not smoothed AGC sample + float sampleAgc = 0.0f; // Smoothed AGC sample + // variables used in effects - uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated) - uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated) - bool samplePeak = 0; // Boolean flag for peak. Responding routine must reset this flag float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc - bool udpSamplePeak = 0; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData - int16_t micIn = 0; // Current sample starts with negative values and large values, which is why it's 16 bit signed - double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controler. - uint32_t timeOfPeak = 0; - unsigned long lastTime = 0; // last time of running UDP Microphone Sync - float micLev = 0.0f; // Used to convert returned value to have '0' as minimum. A leveller - float expAdjF = 0.0f; // Used for exponential filter. - - bool udpSyncConnected = false; - uint16_t audioSyncPort = 11988; - - // used for AGC - uint8_t lastMode = 0; // last known effect mode - int last_soundAgc = -1; - double control_integrated = 0.0; // persistent across calls to agcAvg(); "integrator control" = accumulated error - unsigned long last_update_time = 0; - unsigned long last_kick_time = 0; - uint8_t last_user_inputLevel = 0; - // used to feed "Info" Page unsigned long last_UDPTime = 0; // time of last valid UDP sound sync datapacket float maxSample5sec = 0.0f; // max sample (after AGC) in last 5 seconds @@ -503,6 +537,10 @@ class AudioReactive : public Usermod { static const char UDP_SYNC_HEADER_v1[]; // private methods + + //////////////////// + // Debug support // + //////////////////// void logAudio() { #ifdef MIC_LOGGER @@ -525,7 +563,7 @@ class AudioReactive : public Usermod { #ifdef FFT_SAMPLING_LOG #if 0 - for(int i=0; i<16; i++) { + for(int i=0; i maxVal) maxVal = fftResult[i]; if(fftResult[i] < minVal) minVal = fftResult[i]; } - for(int i = 0; i < 16; i++) { + for(int i = 0; i < NUM_GEQ_CHANNELS; i++) { Serial.print(i); Serial.print(":"); Serial.printf("%04ld ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1)); } @@ -574,6 +612,10 @@ class AudioReactive : public Usermod { } // logAudio() + ////////////////////// + // Audio Processing // + ////////////////////// + /* * A "PI controller" multiplier to automatically adjust sound sensitivity. * @@ -668,7 +710,7 @@ class AudioReactive : public Usermod { last_soundAgc = soundAgc; } // agcAvg() - + // post-processing and filtering of MIC sample (micDataReal) from FFTcode() void getSample() { float sampleAdj; // Gain adjusted sample value @@ -729,24 +771,6 @@ class AudioReactive : public Usermod { if (sampleMax < 0.5f) sampleMax = 0.0f; sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples. - - // Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC - uint16_t MinShowDelay = strip.getMinShowDelay(); - - if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed. - samplePeak = false; - udpSamplePeak = false; - } - //if (userVar1 == 0) samplePeak = 0; - - // Poor man's beat detection by seeing if sample > Average + some value. - if ((maxVol > 0) && (binNum > 1) && (fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) { - // This goes through ALL of the 255 bins - but ignores stupid settings - // Then we got a peak, else we don't. The peak has to time out on its own in order to support UDP sound sync. - samplePeak = true; - timeOfPeak = millis(); - udpSamplePeak = true; - } } // getSample() @@ -781,6 +805,26 @@ class AudioReactive : public Usermod { } + ////////////////////// + // UDP Sound Sync // + ////////////////////// + + // try to establish UDP sound sync connection + void connectUDPSoundSync(void) { + // This function tries to establish a UDP sync connection if needed + // necessary as we also want to transmit in "AP Mode", but the standard "connected()" callback only reacts on STA connection + static unsigned long last_connection_attempt = 0; + + if ((audioSyncPort <= 0) || ((audioSyncEnabled & 0x03) == 0)) return; // Sound Sync not enabled + if (udpSyncConnected) return; // already connected + if (!(apActive || interfacesInited)) return; // neither AP nor other connections availeable + if (millis() - last_connection_attempt < 15000) return; // only try once in 15 seconds + + // if we arrive here, we need a UDP connection but don't have one + last_connection_attempt = millis(); + connected(); // try to start UDP + } + void transmitAudioData() { if (!udpSyncConnected) return; @@ -795,7 +839,7 @@ class AudioReactive : public Usermod { udpSamplePeak = false; // Reset udpSamplePeak after we've transmitted it transmitData.reserved1 = 0; - for (int i = 0; i < 16; i++) { + for (int i = 0; i < NUM_GEQ_CHANNELS; i++) { transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254); } @@ -808,12 +852,10 @@ class AudioReactive : public Usermod { return; } // transmitAudioData() - static bool isValidUdpSyncVersion(const char *header) { return strncmp_P(header, PSTR(UDP_SYNC_HEADER), 6) == 0; } - bool receiveAudioData() // check & process new data. return TRUE in case that new audio data was received. { if (!udpSyncConnected) return false; @@ -839,13 +881,7 @@ class AudioReactive : public Usermod { sampleAgc = volumeSmth; multAgc = 1.0f; - // auto-reset sample peak. Need to do it here, because getSample() is not running - uint16_t MinShowDelay = strip.getMinShowDelay(); - if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed. - samplePeak = false; - udpSamplePeak = false; - } - //if (userVar1 == 0) samplePeak = 0; + autoResetPeak(); // Only change samplePeak IF it's currently false. // If it's true already, then the animation still needs to respond. if (!samplePeak) { @@ -855,7 +891,7 @@ class AudioReactive : public Usermod { } //These values are only available on the ESP32 - for (int i = 0; i < 16; i++) fftResult[i] = receivedPacket->fftResult[i]; + for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i]; my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0f); FFT_Magnitude = my_magnitude; @@ -869,6 +905,10 @@ class AudioReactive : public Usermod { } + ////////////////////// + // usermod functions// + ////////////////////// + public: //Functions called by WLED or other usermods @@ -961,6 +1001,7 @@ class AudioReactive : public Usermod { disableSoundProcessing = true; } + if (enabled) connectUDPSoundSync(); initDone = true; } @@ -971,6 +1012,11 @@ class AudioReactive : public Usermod { */ void connected() { + if (udpSyncConnected) { // clean-up: if open, close old UDP sync connection + udpSyncConnected = false; + fftUdp.stop(); + } + if (audioSyncPort > 0 && (audioSyncEnabled & 0x03)) { #ifndef ESP8266 udpSyncConnected = fftUdp.beginMulticast(IPAddress(239, 0, 0, 1), audioSyncPort); @@ -1067,9 +1113,13 @@ class AudioReactive : public Usermod { if (soundAgc) my_magnitude *= multAgc; if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute - limitSampleDynamics(); // optional - makes volumeSmth very smooth and fluent - } + limitSampleDynamics(); + } // if (!disableSoundProcessing) + autoResetPeak(); // auto-reset sample peak after strip minShowDelay + if (!udpSyncConnected) udpSamplePeak = false; // reset UDP samplePeak while UDP is unconnected + + connectUDPSoundSync(); // ensure we have a connection - if needed // UDP Microphone Sync - receive mode if ((audioSyncEnabled & 0x02) && udpSyncConnected) { @@ -1092,7 +1142,7 @@ class AudioReactive : public Usermod { } #endif - // peak sample from last 5 seconds + // Info Page: keep max sample from last 5 seconds if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) { sampleMaxTimer = millis(); maxSample5sec = (0.15 * maxSample5sec) + 0.85 *((soundAgc) ? sampleAgc : sampleAvg); // reset, and start with some smoothing @@ -1100,6 +1150,7 @@ class AudioReactive : public Usermod { } else { if ((sampleAvg >= 1)) maxSample5sec = fmaxf(maxSample5sec, (soundAgc) ? rawSampleAgc : sampleRaw); // follow maximum volume } + //UDP Microphone Sync - transmit mode if ((audioSyncEnabled & 0x01) && (millis() - lastTime > 20)) { // Only run the transmit code IF we're in Transmit mode @@ -1137,8 +1188,9 @@ class AudioReactive : public Usermod { memset(fftCalc, 0, sizeof(fftCalc)); memset(fftAvg, 0, sizeof(fftAvg)); memset(fftResult, 0, sizeof(fftResult)); - for(int i=(init?0:1); i<16; i+=2) fftResult[i] = 16; // make a tiny pattern + for(int i=(init?0:1); i")); + infoArr.add(F("
")); } } diff --git a/wled00/FX.cpp b/wled00/FX.cpp index b24abe41..30019c9d 100644 --- a/wled00/FX.cpp +++ b/wled00/FX.cpp @@ -6251,7 +6251,7 @@ uint16_t mode_gravcentric(void) { // Gravcentric. By Andrew return FRAMETIME; } // mode_gravcentric() -static const char _data_FX_MODE_GRAVCENTRIC[] PROGMEM = "Gravcentric@Rate of fall,Sensitivity;!;!;ix=128,mp12=2,ssim=0,1d,vo"; // Circle, Beatsin +static const char _data_FX_MODE_GRAVCENTRIC[] PROGMEM = "Gravcentric@Rate of fall,Sensitivity;!;!;ix=128,mp12=3,ssim=0,1d,vo"; // Corner, Beatsin /////////////////////// @@ -6387,7 +6387,7 @@ uint16_t mode_midnoise(void) { // Midnoise. By Andrew Tuline. return FRAMETIME; } // mode_midnoise() -static const char _data_FX_MODE_MIDNOISE[] PROGMEM = "Midnoise@Fade rate,Maximum length;,!;!;ix=128,mp12=2,ssim=0,1d,vo"; // Circle, Beatsin +static const char _data_FX_MODE_MIDNOISE[] PROGMEM = "Midnoise@Fade rate,Maximum length;,!;!;ix=128,mp12=1,ssim=0,1d,vo"; // Bar, Beatsin ////////////////////// @@ -6512,7 +6512,7 @@ uint16_t mode_plasmoid(void) { // Plasmoid. By Andrew Tuline. } float volumeSmth = *(float*) um_data->u_data[0]; - SEGMENT.fadeToBlackBy(64); + SEGMENT.fadeToBlackBy(32); plasmoip->thisphase += beatsin8(6,-4,4); // You can change direction and speed individually. plasmoip->thatphase += beatsin8(7,-4,4); // Two phase values to make a complex pattern. By Andrew Tuline. @@ -6664,7 +6664,8 @@ uint16_t mode_blurz(void) { // Blurz. By Andrew Tuline. SEGENV.aux0 = 0; } - SEGMENT.fade_out(SEGMENT.speed); + int fadeoutDelay = (256 - SEGMENT.speed) / 32; + if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fade_out(SEGMENT.speed); SEGENV.step += FRAMETIME; if (SEGENV.step > SPEED_FORMULA_L) { @@ -6732,7 +6733,9 @@ uint16_t mode_freqmap(void) { // Map FFT_MajorPeak to SEGLEN. float my_magnitude = *(float*) um_data->u_data[5] / 4.0f; if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception) - SEGMENT.fade_out(SEGMENT.speed); + if (SEGENV.call == 0) SEGMENT.fill(BLACK); + int fadeoutDelay = (256 - SEGMENT.speed) / 32; + if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fade_out(SEGMENT.speed); int locn = (log10f((float)FFT_MajorPeak) - 1.78f) * (float)SEGLEN/(MAX_FREQ_LOG10 - 1.78f); // log10 frequency range is from 1.78 to 3.71. Let's scale to SEGLEN. if (locn < 1) locn = 0; // avoid underflow @@ -6747,7 +6750,7 @@ uint16_t mode_freqmap(void) { // Map FFT_MajorPeak to SEGLEN. return FRAMETIME; } // mode_freqmap() -static const char _data_FX_MODE_FREQMAP[] PROGMEM = "Freqmap@Fade rate,Starting color;,!;!;mp12=2,ssim=0,1d,fr"; // Circle, Beatsin +static const char _data_FX_MODE_FREQMAP[] PROGMEM = "Freqmap@Fade rate,Starting color;,!;!;mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin /////////////////////// @@ -6802,7 +6805,7 @@ uint16_t mode_freqmatrix(void) { // Freqmatrix. By Andreas Plesch return FRAMETIME; } // mode_freqmatrix() -static const char _data_FX_MODE_FREQMATRIX[] PROGMEM = "Freqmatrix@Time delay,Sound effect,Low bin,High bin,Sensivity;;;mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin +static const char _data_FX_MODE_FREQMATRIX[] PROGMEM = "Freqmatrix@Time delay,Sound effect,Low bin,High bin,Sensivity;;;mp12=3,ssim=0,1d,fr"; // Corner, Beatsin ////////////////////// @@ -6823,7 +6826,10 @@ uint16_t mode_freqpixels(void) { // Freqpixel. By Andrew Tuline. if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception) uint16_t fadeRate = 2*SEGMENT.speed - SEGMENT.speed*SEGMENT.speed/255; // Get to 255 as quick as you can. - SEGMENT.fade_out(fadeRate); + + if (SEGENV.call == 0) SEGMENT.fill(BLACK); + int fadeoutDelay = (256 - SEGMENT.speed) / 64; + if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fade_out(fadeRate); for (int i=0; i < SEGMENT.intensity/32+1; i++) { uint16_t locn = random16(0,SEGLEN); @@ -6955,7 +6961,7 @@ uint16_t mode_gravfreq(void) { // Gravfreq. By Andrew Tuline. return FRAMETIME; } // mode_gravfreq() -static const char _data_FX_MODE_GRAVFREQ[] PROGMEM = "Gravfreq@Rate of fall,Sensivity;,!;!;ix=128,mp12=2,ssim=0,1d,fr"; // Circle, Beatsin +static const char _data_FX_MODE_GRAVFREQ[] PROGMEM = "Gravfreq@Rate of fall,Sensivity;,!;!;ix=128,mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin ////////////////////// @@ -6969,7 +6975,10 @@ uint16_t mode_noisemove(void) { // Noisemove. By: Andrew Tuli } uint8_t *fftResult = (uint8_t*)um_data->u_data[2]; - SEGMENT.fade_out(224); // Just in case something doesn't get faded. + if (SEGENV.call == 0) SEGMENT.fill(BLACK); + //SEGMENT.fade_out(224); // Just in case something doesn't get faded. + int fadeoutDelay = (256 - SEGMENT.speed) / 96; + if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fadeToBlackBy(4+ SEGMENT.speed/4); uint8_t numBins = map(SEGMENT.intensity,0,255,0,16); // Map slider to fftResult bins. for (int i=0; iu_data[4]; float my_magnitude = *(float*) um_data->u_data[5] / 16.0f; - SEGMENT.fadeToBlackBy(64); // Just in case something doesn't get faded. + if (SEGENV.call == 0) SEGMENT.fill(BLACK); + SEGMENT.fadeToBlackBy(16); // Just in case something doesn't get faded. float frTemp = FFT_MajorPeak; uint8_t octCount = 0; // Octave counter. uint8_t volTemp = 0; - if (my_magnitude > 32) volTemp = 255; // We need to squelch out the background noise. + volTemp = 32.0f + my_magnitude * 1.5f; // brightness = volume (overflows are handled in next lines) + if (my_magnitude < 48) volTemp = 0; // We need to squelch out the background noise. + if (my_magnitude > 144) volTemp = 255; // everything above this is full brightness while ( frTemp > 249 ) { octCount++; // This should go up to 5. @@ -7017,7 +7029,7 @@ uint16_t mode_rocktaves(void) { // Rocktaves. Same note from eac return FRAMETIME; } // mode_rocktaves() -static const char _data_FX_MODE_ROCKTAVES[] PROGMEM = "Rocktaves@;,!;!;mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin +static const char _data_FX_MODE_ROCKTAVES[] PROGMEM = "Rocktaves@;,!;!;mp12=1,ssim=0,1d,fr"; // Bar, Beatsin /////////////////////// @@ -7101,10 +7113,13 @@ uint16_t mode_2DGEQ(void) { // By Will Tatam. Code reduction by Ewoud Wijma. rippleTime = true; } - SEGMENT.fadeToBlackBy(SEGMENT.speed); + if (SEGENV.call == 0) SEGMENT.fill(BLACK); + int fadeoutDelay = (256 - SEGMENT.speed) / 64; + if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fadeToBlackBy(SEGMENT.speed); for (int x=0; x < cols; x++) { uint8_t band = map(x, 0, cols-1, 0, NUM_BANDS - 1); + if (NUM_BANDS < 16) band = map(band, 0, NUM_BANDS - 1, 0, 15); // always use full range. comment out this line to get the previous behaviour. band = constrain(band, 0, 15); uint16_t colorIndex = band * 17; uint16_t barHeight = map(fftResult[band], 0, 255, 0, rows); // do not subtract -1 from rows here