From 899899094f76d07528f525e83d5806c30f328c65 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Mon, 24 Jun 2024 17:01:43 +0200 Subject: [PATCH 01/29] sound sync: use last remaining gap to transmit soundPressure * use last remaining two bytes in audioSyncPacket for transmitting soundPressure * 0x0 is treated as "legacy value" --> soundPressure = volumeSmth; * decodeAudioData: ensure receivedPacket struct members are correctly aligned - strictly speaking it is not safe to cast a uint8_t* as it does not offer any alignment guarantees. * remove 8266 special handling in audioreactive::setup() --- usermods/audioreactive/audio_reactive.h | 88 +++++++++++++------------ 1 file changed, 47 insertions(+), 41 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index b958184c..753f2b54 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -1050,7 +1050,7 @@ class AudioReactive : public Usermod { // new "V2" audiosync struct - 44 Bytes struct __attribute__ ((packed)) audioSyncPacket { // WLEDMM "packed" ensures that there are no additional gaps char header[6]; // 06 Bytes offset 0 - uint8_t gap1[2]; // gap added by compiler: 02 Bytes, offset 6 + uint8_t pressure[2]; // 02 Bytes, offset 6 - sound pressure as fixed point (8bit integer, 8bit fraction) float sampleRaw; // 04 Bytes offset 8 - either "sampleRaw" or "rawSampleAgc" depending on soundAgc setting float sampleSmth; // 04 Bytes offset 12 - either "sampleAvg" or "sampleAgc" depending on soundAgc setting uint8_t samplePeak; // 01 Bytes offset 16 - 0 no peak; >=1 peak detected. In future, this will also provide peak Magnitude @@ -1065,8 +1065,8 @@ class AudioReactive : public Usermod { struct audioSyncPacket_v1 { char header[6]; // 06 Bytes uint8_t myVals[32]; // 32 Bytes - int sampleAgc; // 04 Bytes - int sampleRaw; // 04 Bytes + int32_t sampleAgc; // 04 Bytes + int32_t sampleRaw; // 04 Bytes float sampleAvg; // 04 Bytes bool samplePeak; // 01 Bytes uint8_t fftResult[16]; // 16 Bytes @@ -1602,6 +1602,14 @@ class AudioReactive : public Usermod { transmitData.fftResult[i] = fftResult[i]; } + // WLEDMM transmit soundPressure as 16 bit fixed point + uint32_t pressure16bit = max(0.0f, soundPressure) * 256.0f; // convert to fixed point, remove negative values + uint16_t pressInt = pressure16bit / 256; // integer part + uint16_t pressFract = pressure16bit % 256; // faction part + if (pressInt > 255) pressInt = 255; // saturation at 255 + transmitData.pressure[0] = (uint8_t)pressInt; + transmitData.pressure[1] = (uint8_t)pressFract; + transmitData.FFT_Magnitude = my_magnitude; transmitData.FFT_MajorPeak = FFT_MajorPeak; @@ -1622,30 +1630,33 @@ class AudioReactive : public Usermod { bool decodeAudioData(int packetSize, uint8_t *fftBuff) { if((0 == packetSize) || (nullptr == fftBuff)) return false; // sanity check - audioSyncPacket *receivedPacket = reinterpret_cast(fftBuff); + //audioSyncPacket *receivedPacket = reinterpret_cast(fftBuff); + audioSyncPacket receivedPacket; + memset(&receivedPacket, 0, sizeof(receivedPacket)); // start clean + memcpy(&receivedPacket, fftBuff, min((unsigned)packetSize, (unsigned)sizeof(receivedPacket))); // don't violate alignment - thanks @willmmiles // validate sequence, discard out-of-sequence packets static uint8_t lastFrameCounter = 0; // add info for UI - if ((receivedPacket->frameCounter > 0) && (lastFrameCounter > 0)) receivedFormat = 3; // v2+ + if ((receivedPacket.frameCounter > 0) && (lastFrameCounter > 0)) receivedFormat = 3; // v2+ else receivedFormat = 2; // v2 // check sequence bool sequenceOK = false; - if(receivedPacket->frameCounter > lastFrameCounter) sequenceOK = true; // sequence OK - if((lastFrameCounter < 12) && (receivedPacket->frameCounter > 248)) sequenceOK = false; // prevent sequence "roll-back" due to late packets (1->254) - if((lastFrameCounter > 248) && (receivedPacket->frameCounter < 12)) sequenceOK = true; // handle roll-over (255 -> 0) + if(receivedPacket.frameCounter > lastFrameCounter) sequenceOK = true; // sequence OK + if((lastFrameCounter < 12) && (receivedPacket.frameCounter > 248)) sequenceOK = false; // prevent sequence "roll-back" due to late packets (1->254) + if((lastFrameCounter > 248) && (receivedPacket.frameCounter < 12)) sequenceOK = true; // handle roll-over (255 -> 0) if(audioSyncSequence == false) sequenceOK = true; // sequence checking disabled by user - if((sequenceOK == false) && (receivedPacket->frameCounter != 0)) { // always accept "0" - its the legacy value - DEBUGSR_PRINTF("Skipping audio frame out of order or duplicated - %u vs %u\n", lastFrameCounter, receivedPacket->frameCounter); + if((sequenceOK == false) && (receivedPacket.frameCounter != 0)) { // always accept "0" - its the legacy value + DEBUGSR_PRINTF("Skipping audio frame out of order or duplicated - %u vs %u\n", lastFrameCounter, receivedPacket.frameCounter); return false; // reject out-of sequence frame } else { - lastFrameCounter = receivedPacket->frameCounter; + lastFrameCounter = receivedPacket.frameCounter; } // update samples for effects - volumeSmth = fmaxf(receivedPacket->sampleSmth, 0.0f); - volumeRaw = fmaxf(receivedPacket->sampleRaw, 0.0f); + volumeSmth = fmaxf(receivedPacket.sampleSmth, 0.0f); + volumeRaw = fmaxf(receivedPacket.sampleRaw, 0.0f); #ifdef ARDUINO_ARCH_ESP32 // update internal samples sampleRaw = volumeRaw; @@ -1658,18 +1669,26 @@ class AudioReactive : public Usermod { // If it's true already, then the animation still needs to respond. autoResetPeak(); if (!samplePeak) { - samplePeak = receivedPacket->samplePeak >0 ? true:false; + samplePeak = receivedPacket.samplePeak >0 ? true:false; if (samplePeak) timeOfPeak = millis(); //userVar1 = samplePeak; } //These values are only computed by ESP32 - for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i]; - my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0f); + 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; - FFT_MajorPeak = constrain(receivedPacket->FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects - soundPressure = volumeSmth; // substitute - V2 format does not (yet) include this value - agcSensitivity = 128.0f; // substitute - V2 format does not (yet) include this value - zeroCrossingCount = receivedPacket->zeroCrossingCount; + FFT_MajorPeak = constrain(receivedPacket.FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects + agcSensitivity = 128.0f; // substitute - V2 format does not include this value + zeroCrossingCount = receivedPacket.zeroCrossingCount; + + // WLEDMM extract soundPressure + if ((receivedPacket.pressure[0] != 0) || (receivedPacket.pressure[1] != 0)) { + // found something in gap "reserved2" + soundPressure = float(receivedPacket.pressure[1]) / 256.0f; // fractional part + soundPressure += float(receivedPacket.pressure[0]); // integer part + } else { + soundPressure = volumeSmth; // fallback + } return true; } @@ -1786,36 +1805,23 @@ class AudioReactive : public Usermod { um_data->u_type[4] = UMT_FLOAT; um_data->u_data[5] = &my_magnitude; // used (New) um_data->u_type[5] = UMT_FLOAT; -#ifdef ARDUINO_ARCH_ESP32 um_data->u_data[6] = &maxVol; // assigned in effect function from UI element!!! (Puddlepeak, Ripplepeak, Waterfall) um_data->u_type[6] = UMT_BYTE; um_data->u_data[7] = &binNum; // assigned in effect function from UI element!!! (Puddlepeak, Ripplepeak, Waterfall) um_data->u_type[7] = UMT_BYTE; +#ifdef ARDUINO_ARCH_ESP32 um_data->u_data[8] = &FFT_MajPeakSmth; // new um_data->u_type[8] = UMT_FLOAT; +#else + um_data->u_data[8] = &FFT_MajorPeak; // substitute for 8266 + um_data->u_type[8] = UMT_FLOAT; +#endif um_data->u_data[9] = &soundPressure; // used (New) um_data->u_type[9] = UMT_FLOAT; - um_data->u_data[10] = &agcSensitivity; // used (New) + um_data->u_data[10] = &agcSensitivity; // used (New) - dummy value on 8266 um_data->u_type[10] = UMT_FLOAT; - um_data->u_data[11] = &zeroCrossingCount; + um_data->u_data[11] = &zeroCrossingCount; // for auto playlist usermod um_data->u_type[11] = UMT_UINT16; -#else - // ESP8266 - // See https://github.com/MoonModules/WLED/pull/60#issuecomment-1666972133 for explanation of these alternative sources of data - - um_data->u_data[6] = &maxVol; // assigned in effect function from UI element!!! (Puddlepeak, Ripplepeak, Waterfall) - um_data->u_type[6] = UMT_BYTE; - um_data->u_data[7] = &binNum; // assigned in effect function from UI element!!! (Puddlepeak, Ripplepeak, Waterfall) - um_data->u_type[7] = UMT_BYTE; - um_data->u_data[8] = &FFT_MajorPeak; // new - substitute for FFT_MajPeakSmth - um_data->u_type[8] = UMT_FLOAT; - um_data->u_data[9] = &volumeSmth; // used (New) - substitute for soundPressure - um_data->u_type[9] = UMT_FLOAT; - um_data->u_data[10] = &agcSensitivity; // used (New) - dummy value (128 => 50%) - um_data->u_type[10] = UMT_FLOAT; - um_data->u_data[11] = &zeroCrossingCount; - um_data->u_type[11] = UMT_UINT16; -#endif } #ifdef ARDUINO_ARCH_ESP32 @@ -2201,7 +2207,7 @@ class AudioReactive : public Usermod { volumeSmth =0.0f; volumeRaw =0; my_magnitude = 0.1; FFT_Magnitude = 0.01; FFT_MajorPeak = 2; - soundPressure = 1.0f; + soundPressure = 1.0f; agcSensitivity = 64.0f; #ifdef ARDUINO_ARCH_ESP32 multAgc = 1; From c876267c95b0dd2e29251d69e4c8f9c6f7423027 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Mon, 24 Jun 2024 17:14:09 +0200 Subject: [PATCH 02/29] Update audio_reactive.h - debug fix --- usermods/audioreactive/audio_reactive.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 753f2b54..52f8e17f 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -1981,7 +1981,7 @@ class AudioReactive : public Usermod { USER_PRINTF("\naudioSyncPacket_v1 size = %d\n", sizeof(audioSyncPacket_v1)); // size 88 USER_PRINTF("audioSyncPacket size = %d\n", sizeof(audioSyncPacket)); // size 44 USER_PRINTF("| char header[6] offset = %2d size = %2d\n", offsetof(audioSyncPacket, header[0]), sizeof(data.header)); // offset 0 size 6 - USER_PRINTF("| uint8_t gap1[2] offset = %2d size = %2d\n", offsetof(audioSyncPacket, gap1[0]), sizeof(data.gap1)); // offset 6 size 2 + USER_PRINTF("| uint8_t pressure[2] offset = %2d size = %2d\n", offsetof(audioSyncPacket, pressure[0]), sizeof(data.pressure)); // offset 6 size 2 USER_PRINTF("| float sampleRaw offset = %2d size = %2d\n", offsetof(audioSyncPacket, sampleRaw), sizeof(data.sampleRaw)); // offset 8 size 4 USER_PRINTF("| float sampleSmth offset = %2d size = %2d\n", offsetof(audioSyncPacket, sampleSmth), sizeof(data.sampleSmth)); // offset 12 size 4 USER_PRINTF("| uint8_t samplePeak offset = %2d size = %2d\n", offsetof(audioSyncPacket, samplePeak), sizeof(data.samplePeak)); // offset 16 size 1 From 2510292d1a8eb099f95aa4307405ed1dad6f892c Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Mon, 24 Jun 2024 18:04:18 +0200 Subject: [PATCH 03/29] ws preview: better handling of RGBW avoid over-saturation in preview --- wled00/ws.cpp | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/wled00/ws.cpp b/wled00/ws.cpp index 32420fcc..5b8bda42 100644 --- a/wled00/ws.cpp +++ b/wled00/ws.cpp @@ -277,9 +277,10 @@ static bool sendLiveLedsWs(uint32_t wsClient) // WLEDMM added "static" // WLEDMM begin: preview with color gamma correction if (gammaCorrectPreview) { uint8_t w = W(c); // not sure why, but it looks better if using "white" without corrections - buffer[pos++] = qadd8(w, unGamma8(R(c))); //R, add white channel to RGB channels as a simple RGBW -> RGB map - buffer[pos++] = qadd8(w, unGamma8(G(c))); //G - buffer[pos++] = qadd8(w, unGamma8(B(c))); //B + if (w>0) c = color_add(c, RGBW32(w, w, w, 0), false); // add white channel to RGB channels - color_add() will prevent over-saturation + buffer[pos++] = unGamma8(R(c)); //R + buffer[pos++] = unGamma8(G(c)); //G + buffer[pos++] = unGamma8(B(c)); //B } else { // WLEDMM end uint8_t w = W(c); // WLEDMM small optimization From bc006268ae2209e9c5d24737bc3445b7d8593e7c Mon Sep 17 00:00:00 2001 From: Frank Date: Mon, 24 Jun 2024 21:50:49 +0200 Subject: [PATCH 04/29] new MM build for -C3 with 2MB flash (no OTA) --- platformio.ini | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/platformio.ini b/platformio.ini index 5ada7d47..7d74b2a3 100644 --- a/platformio.ini +++ b/platformio.ini @@ -80,6 +80,7 @@ default_envs = ;; esp32s2_tinyUF2_PSRAM_S ;; experimental - only for adafruit -S2 boards with tinyUF2 bootloader !!! esp32s2_PSRAM_M ;; experimental esp32c3dev_4MB_M ;; experimental + esp32c3dev_2MB_M ;; experimental - 2MB Flash, no OTA esp32c3mini_dio_4MB_M ;; for boards that need "dio" flash mode (instead of qio) seeed_esp32c3_4MB_S ;; experimental esp32_4MB_V4_S ;; experimental @@ -2180,6 +2181,33 @@ build_flags = ${env:esp32c3dev_4MB_M.build_flags} ; RAM: [=== ] 25.8% (used 84700 bytes from 327680 bytes) ; Flash: [==========] 98.7% (used 1552582 bytes from 1572864 bytes) +[env:esp32c3dev_2MB_M] +extends = env:esp32c3dev_4MB_M +board = lolin_c3_mini +;;; replace WLED_RELEASE_NAME, disable CDC_ON_BOOT +build_unflags = ${env:esp32c3dev_4MB_M.build_unflags} + -DARDUINO_USB_CDC_ON_BOOT=1 + -D WLED_RELEASE_NAME=esp32c3dev_4MB_M + +;;; 2MB Flash, no OTA +board_build.partitions = tools/WLED_ESP32_2MB_noOTA.csv +board_build.flash_mode = dio +board_upload.flash_size = 2MB +board_upload.maximum_size = 2097152 + +build_flags = ${env:esp32c3dev_4MB_M.build_flags} + -D WLED_WATCHDOG_TIMEOUT=0 + -D WLED_DISABLE_OTA + ; -DARDUINO_USB_CDC_ON_BOOT=1 ;; for virtual CDC USB + -DARDUINO_USB_CDC_ON_BOOT=0 ;; for serial-to-USB chip + -D WLED_RELEASE_NAME=esp32c3dev_2MB_M + -D WLED_DISABLE_BROWNOUT_DET ;; the board only has a 500mA LDO, better to disable brownout detection + -D WLED_DISABLE_ADALIGHT ;; to disable serial protocols for boards with CDC USB (Serial RX will receive junk commands, unless its pulled down by resistor) + -D HW_PIN_SDA=0 -D HW_PIN_SCL=1 ;; avoid pin conflicts + +; RAM: [=== ] 25.3% (used 82828 bytes from 327680 bytes) +; Flash: [==========] 97.9% (used 1540138 bytes from 1572864 bytes) + ;; MM environment for "seeed xiao -C3" boards [env:seeed_esp32c3_4MB_S] extends = env:esp32c3dev_4MB_M From c67d5dbdc4df3f96d457bde54d61b78a1637ec38 Mon Sep 17 00:00:00 2001 From: Frank Date: Mon, 24 Jun 2024 21:51:34 +0200 Subject: [PATCH 05/29] build 2406240 - release v0.14.1-b32 --- wled00/wled.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/wled00/wled.h b/wled00/wled.h index 2c7910a6..8ff90dd5 100644 --- a/wled00/wled.h +++ b/wled00/wled.h @@ -8,7 +8,7 @@ */ // version code in format yymmddb (b = daily build) -#define VERSION 2406230 +#define VERSION 2406240 // WLEDMM - you can check for this define in usermods, to only enabled WLEDMM specific code in the "right" fork. Its not defined in AC WLED. #define _MoonModules_WLED_ From 2dd9a8232e614336537a9d3734dbb3222e8ef493 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Mon, 24 Jun 2024 22:32:33 +0200 Subject: [PATCH 06/29] kicking gh build re-run --- platformio.ini | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/platformio.ini b/platformio.ini index 7d74b2a3..6ae4c790 100644 --- a/platformio.ini +++ b/platformio.ini @@ -84,7 +84,7 @@ default_envs = esp32c3mini_dio_4MB_M ;; for boards that need "dio" flash mode (instead of qio) seeed_esp32c3_4MB_S ;; experimental esp32_4MB_V4_S ;; experimental - esp32_16MB_V4_S ;; experimental, optimized for speed + esp32_16MB_V4_S ;; experimental - optimized for speed esp32_16MB_V4_M ;; experimental esp32_16MB_V4_M_debug ;; experimental esp32_pico_4MB_V4_S ;; experimental - may work better in case you experience wifi connectivity problems From 93ec9b915116dd6d8efbe5185f9a849e98624b85 Mon Sep 17 00:00:00 2001 From: Troy <5659019+troyhacks@users.noreply.github.com> Date: Tue, 25 Jun 2024 23:28:33 -0400 Subject: [PATCH 07/29] Stop double first pixel on waterfall effect --- wled00/FX.cpp | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/wled00/FX.cpp b/wled00/FX.cpp index d9047e7d..416884cf 100644 --- a/wled00/FX.cpp +++ b/wled00/FX.cpp @@ -7794,13 +7794,14 @@ uint16_t mode_waterfall(void) { // Waterfall. By: Andrew Tulin uint8_t pixCol = (log10f(FFT_MajorPeak) - 2.26f) * 150; // 22Khz sampling - log10 frequency range is from 2.26 (182hz) to 3.967 (9260hz). Let's scale accordingly. if (FFT_MajorPeak < 182.0f) pixCol = 0; // handle underflow + // loop will not execute if SEGLEN equals 1 + for (int i = 0; i < SEGLEN-1; i++) SEGMENT.setPixelColor(i, SEGMENT.getPixelColor(i+1)); // shift left + if (samplePeak) { SEGMENT.setPixelColor(SEGLEN-1, CHSV(92,92,92)); } else { SEGMENT.setPixelColor(SEGLEN-1, color_blend(SEGCOLOR(1), SEGMENT.color_from_palette(pixCol+SEGMENT.intensity, false, PALETTE_SOLID_WRAP, 0), (int)my_magnitude)); } - // loop will not execute if SEGLEN equals 1 - for (int i = 0; i < SEGLEN-1; i++) SEGMENT.setPixelColor(i, SEGMENT.getPixelColor(i+1)); // shift left } return FRAMETIME; From f432cb20dc2c3d0a38d6f12d850baac91ce26e3a Mon Sep 17 00:00:00 2001 From: Frank Date: Mon, 1 Jul 2024 02:14:00 +0200 Subject: [PATCH 08/29] remove experiments:freqRMS option the default value seems to work well. --- usermods/audioreactive/audio_reactive.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 52f8e17f..a0414165 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -237,9 +237,9 @@ static uint8_t useInputFilter = 0; // enables low-cut fil //WLEDMM add experimental settings static uint8_t micLevelMethod = 0; // 0=old "floating" miclev, 1=new "freeze" mode, 2=fast freeze mode (mode 2 may not work for you) #if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3) -static uint8_t averageByRMS = false; // false: use mean value, true: use RMS (root mean squared). use simpler method on slower MCUs. +static constexpr uint8_t averageByRMS = false; // false: use mean value, true: use RMS (root mean squared). use simpler method on slower MCUs. #else -static uint8_t averageByRMS = true; // false: use mean value, true: use RMS (root mean squared). use better method on fast MCUs. +static constexpr uint8_t averageByRMS = true; // false: use mean value, true: use RMS (root mean squared). use better method on fast MCUs. #endif static uint8_t freqDist = 0; // 0=old 1=rightshift mode @@ -436,7 +436,7 @@ static float fftAddAvgRMS(int from, int to) { static float fftAddAvg(int from, int to) { if (from == to) return vReal[from]; // small optimization - if (averageByRMS) return fftAddAvgRMS(from, to); // use SMS + if (averageByRMS) return fftAddAvgRMS(from, to); // use RMS else return fftAddAvgLin(from, to); // use linear average } @@ -2657,7 +2657,7 @@ class AudioReactive : public Usermod { JsonObject poweruser = top.createNestedObject("experiments"); poweruser[F("micLev")] = micLevelMethod; poweruser[F("freqDist")] = freqDist; - poweruser[F("freqRMS")] = averageByRMS; + //poweruser[F("freqRMS")] = averageByRMS; JsonObject freqScale = top.createNestedObject("frequency"); freqScale[F("scale")] = FFTScalingMode; @@ -2729,7 +2729,7 @@ class AudioReactive : public Usermod { //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["experiments"][F("freqRMS")], averageByRMS); configComplete &= getJsonValue(top["frequency"][F("scale")], FFTScalingMode); configComplete &= getJsonValue(top["frequency"][F("profile")], pinkIndex); //WLEDMM @@ -2833,10 +2833,10 @@ class AudioReactive : public Usermod { 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','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);")); From 4f9f250a1d467e3d64002639c85f225e3a09cc70 Mon Sep 17 00:00:00 2001 From: Frank Date: Mon, 1 Jul 2024 02:29:14 +0200 Subject: [PATCH 09/29] only run FFT.dcRemoval() when no filtering was applied FFT.dcRemoval() may introduce unwanted artifacts into the FFT result. --- usermods/audioreactive/audio_reactive.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index a0414165..2543cf34 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -562,12 +562,14 @@ void FFTcode(void * parameter) // band pass filter - can reduce noise floor by a factor of 50 // downside: frequencies below 100Hz will be ignored + bool doDCRemoval = false; // DCRemove is only necessary if we don't use any kind of low-cut filtering if ((useInputFilter > 0) && (useInputFilter < 99)) { switch(useInputFilter) { case 1: runMicFilter(samplesFFT, vReal); break; // PDM microphone bandpass case 2: runDCBlocker(samplesFFT, vReal); break; // generic Low-Cut + DC blocker (~40hz cut-off) + default: doDCRemoval = true; break; } - } + } else doDCRemoval = true; // set imaginary parts to 0 memset(vImag, 0, sizeof(vImag)); @@ -617,7 +619,7 @@ void FFTcode(void * parameter) if ((skipSecondFFT == false) || (isFirstRun == true)) { // run FFT (takes 2-3ms on ESP32, ~12ms on ESP32-S2, ~30ms on -C3) #ifdef UM_AUDIOREACTIVE_USE_NEW_FFT - FFT.dcRemoval(); // remove DC offset + if (doDCRemoval) FFT.dcRemoval(); // remove DC offset #if !defined(FFT_PREFER_EXACT_PEAKS) FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude accuracy #else From f3cbe75506bf4ac022f352d80e78879ade2fea07 Mon Sep 17 00:00:00 2001 From: Frank Date: Mon, 1 Jul 2024 02:39:44 +0200 Subject: [PATCH 10/29] remove support for old ArduinoFFT library < 1.9.2 there is no reason any more to still use the old arduinoFFT library version. --- usermods/audioreactive/audio_reactive.h | 27 ------------------------- usermods/audioreactive/readme.md | 12 ++--------- 2 files changed, 2 insertions(+), 37 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 2543cf34..c466cee5 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -381,7 +381,6 @@ constexpr float binWidth = SAMPLE_RATE / (float)samplesFFT; // frequency range o // Create FFT object -#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT // lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2 #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) // these options actually cause slow-down on -S2 (-S2 doesn't have floating point hardware) @@ -390,13 +389,8 @@ constexpr float binWidth = SAMPLE_RATE / (float)samplesFFT; // frequency range o #endif #define sqrt(x) sqrtf(x) // little hack that reduces FFT time by 10-50% on ESP32 (as alternative to FFT_SQRT_APPROXIMATION) #define sqrt_internal sqrtf // see https://github.com/kosme/arduinoFFT/pull/83 -#else - // around 50% slower on -S2 -// lib_deps += https://github.com/blazoncek/arduinoFFT.git -#endif #include -#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT #if defined(FFT_LIB_REV) && FFT_LIB_REV > 0x19 // arduinoFFT 2.x has a slightly different API static ArduinoFFT FFT = ArduinoFFT( vReal, vImag, samplesFFT, SAMPLE_RATE, true); @@ -405,9 +399,6 @@ constexpr float binWidth = SAMPLE_RATE / (float)samplesFFT; // frequency range o static float windowWeighingFactors[samplesFFT] = {0.0f}; // cache for FFT windowing factors static ArduinoFFT FFT = ArduinoFFT( vReal, vImag, samplesFFT, SAMPLE_RATE, windowWeighingFactors); #endif -#else -static arduinoFFT FFT = arduinoFFT(vReal, vImag, samplesFFT, SAMPLE_RATE); -#endif // Helper functions @@ -618,7 +609,6 @@ void FFTcode(void * parameter) if (fabsf(volumeSmth) > 0.25f) { // noise gate open if ((skipSecondFFT == false) || (isFirstRun == true)) { // run FFT (takes 2-3ms on ESP32, ~12ms on ESP32-S2, ~30ms on -C3) - #ifdef UM_AUDIOREACTIVE_USE_NEW_FFT if (doDCRemoval) FFT.dcRemoval(); // remove DC offset #if !defined(FFT_PREFER_EXACT_PEAKS) FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude accuracy @@ -628,19 +618,6 @@ void FFTcode(void * parameter) FFT.compute( FFTDirection::Forward ); // Compute FFT FFT.complexToMagnitude(); // Compute magnitudes vReal[0] = 0; // The remaining DC offset on the signal produces a strong spike on position 0 that should be eliminated to avoid issues. - #else - FFT.DCRemoval(); // let FFT lib remove DC component, so we don't need to care about this in getSamples() - - //FFT.Windowing( FFT_WIN_TYP_HAMMING, FFT_FORWARD ); // Weigh data - standard Hamming window - //FFT.Windowing( FFT_WIN_TYP_BLACKMAN, FFT_FORWARD ); // Blackman window - better side freq rejection - #if !defined(FFT_PREFER_EXACT_PEAKS) - FFT.Windowing( FFT_WIN_TYP_FLT_TOP, FFT_FORWARD ); // Flat Top Window - better amplitude accuracy - #else - FFT.Windowing( FFT_WIN_TYP_BLACKMAN_HARRIS, FFT_FORWARD );// Blackman-Harris - excellent sideband rejection - #endif - FFT.Compute( FFT_FORWARD ); // Compute FFT - FFT.ComplexToMagnitude(); // Compute magnitudes - #endif float last_majorpeak = FFT_MajorPeak; float last_magnitude = FFT_Magnitude; @@ -651,16 +628,12 @@ void FFTcode(void * parameter) vReal[binInd] *= pinkFactors[binInd]; #endif - #ifdef UM_AUDIOREACTIVE_USE_NEW_FFT #if defined(FFT_LIB_REV) && FFT_LIB_REV > 0x19 // arduinoFFT 2.x has a slightly different API FFT.majorPeak(&FFT_MajorPeak, &FFT_Magnitude); #else FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant #endif - #else - FFT.MajorPeak(&FFT_MajorPeak, &FFT_Magnitude); - #endif if (FFT_MajorPeak < (SAMPLE_RATE / samplesFFT)) {FFT_MajorPeak = 1.0f; FFT_Magnitude = 0;} // too low - use zero if (FFT_MajorPeak > (0.42f * SAMPLE_RATE)) {FFT_MajorPeak = last_majorpeak; FFT_Magnitude = last_magnitude;} // too high - keep last peak diff --git a/usermods/audioreactive/readme.md b/usermods/audioreactive/readme.md index 47804b61..a80fa681 100644 --- a/usermods/audioreactive/readme.md +++ b/usermods/audioreactive/readme.md @@ -27,17 +27,9 @@ Currently ESP8266 is not supported, due to low speed and small RAM of this chip. There are however plans to create a lightweight audioreactive for the 8266, with reduced features. ## Installation -### using customised _arduinoFFT_ library for use with this usermod -Add `-D USERMOD_AUDIOREACTIVE` to your PlatformIO environment `build_flags`, as well as `https://github.com/blazoncek/arduinoFFT.git` to your `lib_deps`. -If you are not using PlatformIO (which you should) try adding `#define USERMOD_AUDIOREACTIVE` to *my_config.h* and make sure you have _arduinoFFT_ library downloaded and installed. +### using latest _arduinoFFT_ library -Customised _arduinoFFT_ library for use with this usermod can be found at https://github.com/blazoncek/arduinoFFT.git - -### using latest (develop) _arduinoFFT_ library -Alternatively, you can use the latest arduinoFFT development version. -ArduinoFFT `develop` library is slightly more accurate, and slightly faster than our customised library, however also needs additional 2kB RAM. - -* `build_flags` = `-D USERMOD_AUDIOREACTIVE` `-D UM_AUDIOREACTIVE_USE_NEW_FFT` +* `build_flags` = `-D USERMOD_AUDIOREACTIVE` * `lib_deps`= `https://github.com/kosme/arduinoFFT#develop @ 1.9.2` ## Configuration From 9803cecee2b8c3c1559f586f3cc6fcfa2a45a1ab Mon Sep 17 00:00:00 2001 From: Frank Date: Mon, 1 Jul 2024 02:46:14 +0200 Subject: [PATCH 11/29] new idea to reduce stack buffer usage the JS string can be shortened, by putting the usermod into a variable `ux` that is used instead of repeating the string 'Aduioreactive'. For now its just an experiment, to see if the idea works on several browsers. --- usermods/audioreactive/audio_reactive.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index c466cee5..1a73fd49 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -2723,7 +2723,8 @@ class AudioReactive : public Usermod { void appendConfigData() { - oappend(SET_F("addInfo('AudioReactive:help',0,'');")); + oappend(SET_F("ux='AudioReactive';")); // fingers crossed that "ux" isn't already used as JS var, html post parameter or css style + oappend(SET_F("addInfo(ux+':help',0,'');")); #ifdef ARDUINO_ARCH_ESP32 //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 From 2987f0d0459cd7961333fdc2ef57007b08bc0d52 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Mon, 1 Jul 2024 15:46:51 +0200 Subject: [PATCH 12/29] SR_STATS: filter time adding "Filtering Time" to SR statistics (info page) --- usermods/audioreactive/audio_reactive.h | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 1a73fd49..fdacaefc 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -330,6 +330,7 @@ static float FFT_MajPeakSmth = 1.0f; // FFT: (peak) frequency, smooth static float fftTaskCycle = 0; // avg cycle time for FFT task static float fftTime = 0; // avg time for single FFT static float sampleTime = 0; // avg (blocked) time for reading I2S samples +static float filterTime = 0; // avg time for filtering I2S samples #endif // FFT Task variables (filtering and post-processing) @@ -525,7 +526,7 @@ void FFTcode(void * parameter) uint64_t sampleTimeInMillis = (esp_timer_get_time() - start +5ULL) / 10ULL; // "+5" to ensure proper rounding sampleTime = (sampleTimeInMillis*3 + sampleTime*7)/10.0; // smooth } - start = esp_timer_get_time(); // start measuring FFT time + start = esp_timer_get_time(); // start measuring filter time #endif xLastWakeTime = xTaskGetTickCount(); // update "last unblocked time" for vTaskDelay @@ -562,6 +563,15 @@ void FFTcode(void * parameter) } } else doDCRemoval = true; +#if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS) + // timing measurement + if (start < esp_timer_get_time()) { // filter out overflows + uint64_t filterTimeInMillis = (esp_timer_get_time() - start +5ULL) / 10ULL; // "+5" to ensure proper rounding + filterTime = (filterTimeInMillis*3 + filterTime*7)/10.0; // smooth + } + start = esp_timer_get_time(); // start measuring FFT time +#endif + // set imaginary parts to 0 memset(vImag, 0, sizeof(vImag)); @@ -2246,7 +2256,7 @@ class AudioReactive : public Usermod { void onUpdateBegin(bool init) { #ifdef WLED_DEBUG - fftTime = sampleTime = 0; + fftTime = sampleTime filterTime = 0; #endif // gracefully suspend FFT task (if running) disableSoundProcessing = true; @@ -2507,17 +2517,22 @@ class AudioReactive : public Usermod { infoArr.add(roundf(sampleTime)/100.0f); infoArr.add(" ms"); + infoArr = user.createNestedArray(F("Filtering time")); + infoArr.add(roundf(filterTime)/100.0f); + infoArr.add(" ms"); + infoArr = user.createNestedArray(F("FFT time")); infoArr.add(roundf(fftTime)/100.0f); if ((fftTime/100) >= FFT_MIN_CYCLE) // FFT time over budget -> I2S buffer will overflow infoArr.add("! ms"); - else if ((fftTime/80 + sampleTime/80) >= FFT_MIN_CYCLE) // FFT time >75% of budget -> risk of instability + else if ((fftTime/85 + filterTime/85 + sampleTime/85) >= FFT_MIN_CYCLE) // FFT time >75% of budget -> risk of instability infoArr.add(" ms!"); else infoArr.add(" ms"); DEBUGSR_PRINTF("AR I2S cycle time: %5.2f ms\n", roundf(fftTaskCycle)/100.0f); DEBUGSR_PRINTF("AR Sampling time : %5.2f ms\n", roundf(sampleTime)/100.0f); + DEBUGSR_PRINTF("AR filter time : %5.2f ms\n", roundf(filterTime)/100.0f); DEBUGSR_PRINTF("AR FFT time : %5.2f ms\n", roundf(fftTime)/100.0f); #endif #endif From 61afb26d21972488b68c5110853ca8d819dbe993 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Mon, 1 Jul 2024 15:53:20 +0200 Subject: [PATCH 13/29] Update audio_reactive.h - fix compile error --- usermods/audioreactive/audio_reactive.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index fdacaefc..d9247f37 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -2256,7 +2256,7 @@ class AudioReactive : public Usermod { void onUpdateBegin(bool init) { #ifdef WLED_DEBUG - fftTime = sampleTime filterTime = 0; + fftTime = sampleTime = filterTime = 0; #endif // gracefully suspend FFT task (if running) disableSoundProcessing = true; From e98858c751a9e29d59f6a1f46bd94acb2f76749b Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Mon, 1 Jul 2024 16:29:09 +0200 Subject: [PATCH 14/29] audio FASTPATH, part 2 * introducing sliding window FFT, which effectively doubles the rate of samples - and FFT results - produced per second. As a side-effect, it also makes FFT a bit less noisy. As sliding window FFT requires double the number of FFT runs, the feature is only enabled on esp32 and esp32-S3. --- usermods/audioreactive/audio_reactive.h | 107 ++++++++++++++++++++++-- 1 file changed, 98 insertions(+), 9 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index d9247f37..99e80f04 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -41,6 +41,12 @@ * .... */ + +#if defined(WLEDMM_FASTPATH) && defined(CONFIG_IDF_TARGET_ESP32S3) || defined(CONFIG_IDF_TARGET_ESP32) +#define FFT_USE_SLIDING_WINDOW // perform FFT with sliding window = 50% overlap +#endif + + #define FFT_PREFER_EXACT_PEAKS // use different FFT windowing -> results in "sharper" peaks and less "leaking" into other frequencies //#define SR_STATS @@ -172,8 +178,13 @@ static bool limiterOn = false; // bool: enable / disable dynamic #else static bool limiterOn = true; #endif +#ifdef FFT_USE_SLIDING_WINDOW +static uint16_t attackTime = 14; // int: attack time in milliseconds. Default 0.014sec +static uint16_t decayTime = 250; // int: decay time in milliseconds. New default 250ms. +#else static uint16_t attackTime = 50; // int: attack time in milliseconds. Default 0.08sec static uint16_t decayTime = 300; // int: decay time in milliseconds. New default 300ms. Old default was 1.40sec +#endif // peak detection #ifdef ARDUINO_ARCH_ESP32 @@ -242,7 +253,9 @@ static constexpr uint8_t averageByRMS = false; // false: us static constexpr uint8_t averageByRMS = true; // false: use mean value, true: use RMS (root mean squared). use better method on fast MCUs. #endif static uint8_t freqDist = 0; // 0=old 1=rightshift mode - +#ifdef FFT_USE_SLIDING_WINDOW +static uint8_t doSlidingFFT = 1; // 1 = use sliding window FFT (faster & more accurate) +#endif // variables used in effects //static int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc @@ -345,7 +358,11 @@ constexpr SRate_t SAMPLE_RATE = 22050; // Base sample rate in Hz - 22Khz #ifndef WLEDMM_FASTPATH #define FFT_MIN_CYCLE 21 // minimum time before FFT task is repeated. Use with 22Khz sampling #else -#define FFT_MIN_CYCLE 15 // reduce min time, to allow faster catch-up when I2S is lagging + #ifdef FFT_USE_SLIDING_WINDOW + #define FFT_MIN_CYCLE 8 // we only have 12ms to take 1/2 batch of samples + #else + #define FFT_MIN_CYCLE 15 // reduce min time, to allow faster catch-up when I2S is lagging + #endif #endif //#define FFT_MIN_CYCLE 30 // Use with 16Khz sampling //#define FFT_MIN_CYCLE 23 // minimum time before FFT task is repeated. Use with 20Khz sampling @@ -363,7 +380,7 @@ constexpr SRate_t SAMPLE_RATE = 18000; // 18Khz; Physical sample time - // 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. +constexpr uint16_t samplesFFT_2 = 256; // meaningful part of FFT results - only the "lower half" contains useful information. // 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 @@ -473,6 +490,12 @@ void FFTcode(void * parameter) const TickType_t xFrequencyDouble = FFT_MIN_CYCLE * portTICK_PERIOD_MS * 2; static bool isFirstRun = false; +#ifdef FFT_USE_SLIDING_WINDOW + static float oldSamples[samplesFFT_2] = {0.0f}; // previous 50% of samples + static bool haveOldSamples = false; // for sliding window FFT + bool usingOldSamples = false; +#endif + #ifdef FFT_MAJORPEAK_HUMAN_EAR // pre-compute pink noise scaling table for(uint_fast16_t binInd = 0; binInd < samplesFFT; binInd++) { @@ -492,6 +515,9 @@ void FFTcode(void * parameter) // Don't run FFT computing code if we're in Receive mode or in realtime mode if (disableSoundProcessing || (audioSyncEnabled == AUDIOSYNC_REC)) { isFirstRun = false; + #ifdef FFT_USE_SLIDING_WINDOW + haveOldSamples = false; + #endif vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers continue; } @@ -511,7 +537,26 @@ void FFTcode(void * parameter) #endif // get a fresh batch of samples from I2S + memset(vReal, 0, sizeof(vReal)); // start clean +#ifdef FFT_USE_SLIDING_WINDOW + uint16_t readOffset; + if (haveOldSamples && (doSlidingFFT > 0)) { + memcpy(vReal, oldSamples, sizeof(float) * samplesFFT_2); // copy first 50% from buffer + usingOldSamples = true; + readOffset = samplesFFT_2; + } else { + usingOldSamples = false; + readOffset = 0; + } + // read fresh samples, in chunks of 50% + do { + // this looks a bit cumbersome, but it onlyworks this way - any second instance of the getSamples() call delivers junk data. + if (audioSource) audioSource->getSamples(vReal+readOffset, samplesFFT_2); + readOffset += samplesFFT_2; + } while (readOffset < samplesFFT); +#else if (audioSource) audioSource->getSamples(vReal, samplesFFT); +#endif #if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS) // debug info in case that stack usage changes @@ -552,13 +597,23 @@ void FFTcode(void * parameter) if (strip.isServicing()) delay(2); #endif + // normal mode: filter everything + float *samplesStart = vReal; + uint16_t sampleCount = samplesFFT; + #ifdef FFT_USE_SLIDING_WINDOW + if (usingOldSamples) { + // sliding window mode: only latest 50% need filtering + samplesStart = vReal + samplesFFT_2; + sampleCount = samplesFFT_2; + } + #endif // band pass filter - can reduce noise floor by a factor of 50 // downside: frequencies below 100Hz will be ignored bool doDCRemoval = false; // DCRemove is only necessary if we don't use any kind of low-cut filtering if ((useInputFilter > 0) && (useInputFilter < 99)) { switch(useInputFilter) { - case 1: runMicFilter(samplesFFT, vReal); break; // PDM microphone bandpass - case 2: runDCBlocker(samplesFFT, vReal); break; // generic Low-Cut + DC blocker (~40hz cut-off) + case 1: runMicFilter(sampleCount, samplesStart); break; // PDM microphone bandpass + case 2: runDCBlocker(sampleCount, samplesStart); break; // generic Low-Cut + DC blocker (~40hz cut-off) default: doDCRemoval = true; break; } } else doDCRemoval = true; @@ -575,14 +630,24 @@ void FFTcode(void * parameter) // set imaginary parts to 0 memset(vImag, 0, sizeof(vImag)); + #ifdef FFT_USE_SLIDING_WINDOW + memcpy(oldSamples, vReal+samplesFFT_2, sizeof(float) * samplesFFT_2); // copy last 50% to buffer (for sliding window FFT) + haveOldSamples = true; + #endif + // find highest sample in the batch, and count zero crossings float maxSample = 0.0f; // max sample from FFT batch uint_fast16_t newZeroCrossingCount = 0; for (int i=0; i < samplesFFT; i++) { // pick our our current mic sample - we take the max value from all samples that go into FFT - if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) //skip extreme values - normally these are artefacts + if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) { //skip extreme values - normally these are artefacts + #ifdef FFT_USE_SLIDING_WINDOW + if (usingOldSamples) { + if ((i >= samplesFFT_2) && (fabsf(vReal[i]) > maxSample)) maxSample = fabsf(vReal[i]); // only look at newest 50% + } else + #endif if (fabsf((float)vReal[i]) > maxSample) maxSample = fabsf((float)vReal[i]); - + } // WLED-MM/TroyHacks: Calculate zero crossings // if (i < (samplesFFT-1)) { @@ -812,6 +877,11 @@ void FFTcode(void * parameter) if ((audioSource == nullptr) || (audioSource->getType() != AudioSource::Type_I2SAdc)) // the "delay trick" does not help for analog ADC #endif { + #ifdef FFT_USE_SLIDING_WINDOW + if (!usingOldSamples) { + vTaskDelayUntil( &xLastWakeTime, xFrequencyDouble); // we need a double wait when no old data was used + } else + #endif if ((skipSecondFFT == false) || (fabsf(volumeSmth) < 0.25f)) { vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers } else if (isFirstRun == true) { @@ -2523,9 +2593,15 @@ class AudioReactive : public Usermod { infoArr = user.createNestedArray(F("FFT time")); infoArr.add(roundf(fftTime)/100.0f); - if ((fftTime/100) >= FFT_MIN_CYCLE) // FFT time over budget -> I2S buffer will overflow + +#ifdef FFT_USE_SLIDING_WINDOW + unsigned timeBudget = doSlidingFFT ? (FFT_MIN_CYCLE) : fftTaskCycle / 115; +#else + unsigned timeBudget = (FFT_MIN_CYCLE); +#endif + if ((fftTime/100) >= timeBudget) // FFT time over budget -> I2S buffer will overflow infoArr.add("! ms"); - else if ((fftTime/85 + filterTime/85 + sampleTime/85) >= FFT_MIN_CYCLE) // FFT time >75% of budget -> risk of instability + else if ((fftTime/85 + filterTime/85 + sampleTime/85) >= timeBudget) // FFT time >75% of budget -> risk of instability infoArr.add(" ms!"); else infoArr.add(" ms"); @@ -2649,6 +2725,9 @@ class AudioReactive : public Usermod { poweruser[F("freqDist")] = freqDist; //poweruser[F("freqRMS")] = averageByRMS; +#ifdef FFT_USE_SLIDING_WINDOW + poweruser[F("I2S_FastPath")] = doSlidingFFT; +#endif JsonObject freqScale = top.createNestedObject("frequency"); freqScale[F("scale")] = FFTScalingMode; freqScale[F("profile")] = pinkIndex; //WLEDMM @@ -2720,6 +2799,9 @@ class AudioReactive : public Usermod { configComplete &= getJsonValue(top["experiments"][F("micLev")], micLevelMethod); configComplete &= getJsonValue(top["experiments"][F("freqDist")], freqDist); //configComplete &= getJsonValue(top["experiments"][F("freqRMS")], averageByRMS); +#ifdef FFT_USE_SLIDING_WINDOW + configComplete &= getJsonValue(top["experiments"][F("I2S_FastPath")], doSlidingFFT); +#endif configComplete &= getJsonValue(top["frequency"][F("scale")], FFTScalingMode); configComplete &= getJsonValue(top["frequency"][F("profile")], pinkIndex); //WLEDMM @@ -2829,6 +2911,13 @@ class AudioReactive : public Usermod { //oappend(SET_F("addOption(dd,'On',1);")); //oappend(SET_F("addInfo('AudioReactive:experiments:freqRMS',1,'☾');")); +#ifdef FFT_USE_SLIDING_WINDOW + oappend(SET_F("dd=addDropdown(ux,'experiments:I2S_FastPath');")); + oappend(SET_F("addOption(dd,'Off',0);")); + oappend(SET_F("addOption(dd,'On (⎌)',1);")); + oappend(SET_F("addInfo(ux+':experiments:I2S_FastPath',1,'☾');")); +#endif + oappend(SET_F("dd=addDropdown('AudioReactive','dynamics:limiter');")); oappend(SET_F("addOption(dd,'Off',0);")); oappend(SET_F("addOption(dd,'On',1);")); From a317af42fc991348d47aafa173fd89f4d1e3ee33 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Mon, 1 Jul 2024 16:37:44 +0200 Subject: [PATCH 15/29] trying to make effects respond to changes quicker in fastpath mode filter parameter tuning --- usermods/audioreactive/audio_reactive.h | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 99e80f04..6f94415f 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -239,7 +239,11 @@ const double agcFollowFast[AGC_NUM_PRESETS] = { 1/192.f, 1/128.f, 1/256.f}; // const double agcFollowSlow[AGC_NUM_PRESETS] = {1/6144.f,1/4096.f,1/8192.f}; // slowly follow setpoint - ~2-15 secs const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter const double agcControlKi[AGC_NUM_PRESETS] = { 1.7f, 1.85f, 1.2f}; // AGC - PI control, integral gain parameter +#if defined(WLEDMM_FASTPATH) +const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/6.f, 1/5.f, 1/10.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value) +#else const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value) +#endif // AGC presets end static AudioSource *audioSource = nullptr; @@ -594,7 +598,9 @@ void FFTcode(void * parameter) #if defined(WLEDMM_FASTPATH) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && defined(ARDUINO_ARCH_ESP32) // experimental - be nice to LED update task (trying to avoid flickering) - dual core only - if (strip.isServicing()) delay(2); +#if FFTTASK_PRIORITY > 1 + if (strip.isServicing()) delay(1); +#endif #endif // normal mode: filter everything @@ -1364,7 +1370,11 @@ class AudioReactive : public Usermod { // update global vars ONCE - multAgc, sampleAGC, rawSampleAgc multAgc = multAgcTemp; +#if defined(WLEDMM_FASTPATH) + rawSampleAgc = 0.65f * tmpAgc + 0.35f * (float)rawSampleAgc; +#else rawSampleAgc = 0.8f * tmpAgc + 0.2f * (float)rawSampleAgc; +#endif // update smoothed AGC sample if (fabsf(tmpAgc) < 1.0f) sampleAgc = 0.5f * tmpAgc + 0.5f * sampleAgc; // fast path to zero @@ -1483,7 +1493,11 @@ class AudioReactive : public Usermod { } if (sampleMax < 0.5f) sampleMax = 0.0f; +#if defined(WLEDMM_FASTPATH) + sampleAvg = ((sampleAvg * 7.0f) + sampleAdj) / 8.0f; // make reactions a bit more "crisp" in fastpath mode +#else sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples. +#endif sampleAvg = fabsf(sampleAvg); // make sure we have a positive value } // getSample() From 31a12f5098f4fd80e7e7c2ed5c8db67e22017e89 Mon Sep 17 00:00:00 2001 From: Frank Date: Mon, 1 Jul 2024 21:10:45 +0200 Subject: [PATCH 16/29] AudioReactive -> ux before: String buffer usage: 3530 of 4037 bytes After: String buffer usage: 3183 of 4037 bytes --- usermods/audioreactive/audio_reactive.h | 86 ++++++++++++------------- 1 file changed, 43 insertions(+), 43 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index d9247f37..10748b57 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -2744,12 +2744,12 @@ class AudioReactive : public Usermod { //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(");"); + oappend(SET_F("xOpt(ux+':analogmic:pin',1,' ⎌',")); oappendi(AUDIOPIN); oappend(");"); #endif - oappend(SET_F("aOpt('AudioReactive:analogmic:pin',1);")); //only analog options + oappend(SET_F("aOpt(ux+':analogmic:pin',1);")); //only analog options #endif - oappend(SET_F("dd=addDropdown('AudioReactive','digitalmic:type');")); + oappend(SET_F("dd=addDropdown(ux,'digitalmic:type');")); #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) #if SR_DMTYPE==0 oappend(SET_F("addOption(dd,'Generic Analog (⎌)',0);")); @@ -2800,50 +2800,50 @@ class AudioReactive : public Usermod { oappend(SET_F("addOption(dd,'WM8978 ☾',7);")); #endif #ifdef SR_SQUELCH - oappend(SET_F("addInfo('AudioReactive:config:squelch',1,'⎌ ")); oappendi(SR_SQUELCH); oappend("');"); // 0 is field type, 1 is actual field + oappend(SET_F("addInfo(ux+':config:squelch',1,'⎌ ")); oappendi(SR_SQUELCH); oappend("');"); // 0 is field type, 1 is actual field #endif #ifdef SR_GAIN - oappend(SET_F("addInfo('AudioReactive:config:gain',1,'⎌ ")); oappendi(SR_GAIN); oappend("');"); // 0 is field type, 1 is actual field + oappend(SET_F("addInfo(ux+':config:gain',1,'⎌ ")); oappendi(SR_GAIN); oappend("');"); // 0 is field type, 1 is actual field #endif - oappend(SET_F("dd=addDropdown('AudioReactive','config:AGC');")); + oappend(SET_F("dd=addDropdown(ux,'config:AGC');")); oappend(SET_F("addOption(dd,'Off',0);")); oappend(SET_F("addOption(dd,'Normal',1);")); 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("dd=addDropdown(ux,'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("addInfo(ux+':experiments:micLev',1,'☾');")); - oappend(SET_F("dd=addDropdown('AudioReactive','experiments:freqDist');")); + oappend(SET_F("dd=addDropdown(ux,'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("addInfo(ux+':experiments:freqDist',1,'☾');")); - //oappend(SET_F("dd=addDropdown('AudioReactive','experiments:freqRMS');")); + //oappend(SET_F("dd=addDropdown(ux,'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("addInfo(ux+':experiments:freqRMS',1,'☾');")); - oappend(SET_F("dd=addDropdown('AudioReactive','dynamics:limiter');")); + oappend(SET_F("dd=addDropdown(ux,'dynamics:limiter');")); oappend(SET_F("addOption(dd,'Off',0);")); oappend(SET_F("addOption(dd,'On',1);")); - oappend(SET_F("addInfo('AudioReactive:dynamics:limiter',0,' On ');")); // 0 is field type, 1 is actual field - oappend(SET_F("addInfo('AudioReactive:dynamics:rise',1,'ms (♪ effects only)');")); - oappend(SET_F("addInfo('AudioReactive:dynamics:fall',1,'ms (♪ effects only)');")); + oappend(SET_F("addInfo(ux+':dynamics:limiter',0,' On ');")); // 0 is field type, 1 is actual field + oappend(SET_F("addInfo(ux+':dynamics:rise',1,'ms (♪ effects only)');")); + oappend(SET_F("addInfo(ux+':dynamics:fall',1,'ms (♪ effects only)');")); - oappend(SET_F("dd=addDropdown('AudioReactive','frequency:scale');")); + oappend(SET_F("dd=addDropdown(ux,'frequency:scale');")); oappend(SET_F("addOption(dd,'None',0);")); oappend(SET_F("addOption(dd,'Linear (Amplitude)',2);")); oappend(SET_F("addOption(dd,'Square Root (Energy)',3);")); oappend(SET_F("addOption(dd,'Logarithmic (Loudness)',1);")); //WLEDMM add defaults - oappend(SET_F("dd=addDropdown('AudioReactive','frequency:profile');")); + oappend(SET_F("dd=addDropdown(ux,'frequency:profile');")); #if SR_FREQ_PROF==0 oappend(SET_F("addOption(dd,'Generic Microphone (⎌)',0);")); #else @@ -2899,9 +2899,9 @@ 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("addInfo(ux+':frequency:profile',1,'☾');")); #endif - oappend(SET_F("dd=addDropdown('AudioReactive','sync:mode');")); + oappend(SET_F("dd=addDropdown(ux,'sync:mode');")); oappend(SET_F("addOption(dd,'Off',0);")); // AUDIOSYNC_NONE #ifdef ARDUINO_ARCH_ESP32 oappend(SET_F("addOption(dd,'Send',1);")); // AUDIOSYNC_SEND @@ -2911,53 +2911,53 @@ class AudioReactive : public Usermod { oappend(SET_F("addOption(dd,'Receive or Local',6);")); // AUDIOSYNC_REC_PLUS #endif // check_sequence: Receiver skips out-of-sequence packets when enabled - oappend(SET_F("dd=addDropdown('AudioReactive','sync:check_sequence');")); + oappend(SET_F("dd=addDropdown(ux,'sync:check_sequence');")); oappend(SET_F("addOption(dd,'Off',0);")); oappend(SET_F("addOption(dd,'On',1);")); - oappend(SET_F("addInfo('AudioReactive:sync:check_sequence',1,'when receiving ☾
Sync audio data with other WLEDs');")); // must append this to the last field of 'sync' + oappend(SET_F("addInfo(ux+':sync:check_sequence',1,'when receiving ☾
Sync audio data with other WLEDs');")); // must append this to the last field of 'sync' - oappend(SET_F("addInfo('AudioReactive:digitalmic:type',1,'requires reboot!');")); // 0 is field type, 1 is actual field + oappend(SET_F("addInfo(ux+':digitalmic:type',1,'requires reboot!');")); // 0 is field type, 1 is actual field #ifdef ARDUINO_ARCH_ESP32 - oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',0,'sd/data/dout','I2S SD');")); + oappend(SET_F("addInfo(ux+':digitalmic:pin[]',0,'sd/data/dout','I2S SD');")); #ifdef I2S_SDPIN - oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',0,' ⎌',")); oappendi(I2S_SDPIN); oappend(");"); + oappend(SET_F("xOpt(ux+':digitalmic:pin[]',0,' ⎌',")); oappendi(I2S_SDPIN); oappend(");"); #endif - oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',1,'ws/clk/lrck','I2S WS');")); - oappend(SET_F("dRO('AudioReactive:digitalmic:pin[]',1);")); // disable read only pins + oappend(SET_F("addInfo(ux+':digitalmic:pin[]',1,'ws/clk/lrck','I2S WS');")); + oappend(SET_F("dRO(ux+':digitalmic:pin[]',1);")); // disable read only pins #ifdef I2S_WSPIN - oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',1,' ⎌',")); oappendi(I2S_WSPIN); oappend(");"); + oappend(SET_F("xOpt(ux+':digitalmic:pin[]',1,' ⎌',")); oappendi(I2S_WSPIN); oappend(");"); #endif - oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',2,'sck/bclk','I2S SCK');")); - oappend(SET_F("dRO('AudioReactive:digitalmic:pin[]',2);")); // disable read only pins + oappend(SET_F("addInfo(ux+':digitalmic:pin[]',2,'sck/bclk','I2S SCK');")); + oappend(SET_F("dRO(ux+':digitalmic:pin[]',2);")); // disable read only pins #ifdef I2S_CKPIN - oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',2,' ⎌',")); oappendi(I2S_CKPIN); oappend(");"); + oappend(SET_F("xOpt(ux+':digitalmic:pin[]',2,' ⎌',")); oappendi(I2S_CKPIN); oappend(");"); #endif - oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',3,'master clock','I2S MCLK');")); - oappend(SET_F("dRO('AudioReactive:digitalmic:pin[]',3);")); // disable read only pins + oappend(SET_F("addInfo(ux+':digitalmic:pin[]',3,'master clock','I2S MCLK');")); + oappend(SET_F("dRO(ux+':digitalmic:pin[]',3);")); // disable read only pins #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) - oappend(SET_F("dOpt('AudioReactive:digitalmic:pin[]',3,2,2);")); //only use -1, 0, 1 or 3 - oappend(SET_F("dOpt('AudioReactive:digitalmic:pin[]',3,4,39);")); //only use -1, 0, 1 or 3 + oappend(SET_F("dOpt(ux+':digitalmic:pin[]',3,2,2);")); //only use -1, 0, 1 or 3 + oappend(SET_F("dOpt(ux+':digitalmic:pin[]',3,4,39);")); //only use -1, 0, 1 or 3 #endif #ifdef MCLK_PIN - oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',3,' ⎌',")); oappendi(MCLK_PIN); oappend(");"); + oappend(SET_F("xOpt(ux+':digitalmic:pin[]',3,' ⎌',")); oappendi(MCLK_PIN); oappend(");"); #endif - oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',4,'','I2C SDA');")); - oappend(SET_F("rOpt('AudioReactive:digitalmic:pin[]',4,'use global (")); oappendi(i2c_sda); oappend(")',-1);"); + oappend(SET_F("addInfo(ux+':digitalmic:pin[]',4,'','I2C SDA');")); + oappend(SET_F("rOpt(ux+':digitalmic:pin[]',4,'use global (")); oappendi(i2c_sda); oappend(")',-1);"); #ifdef ES7243_SDAPIN - oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',4,' ⎌',")); oappendi(ES7243_SDAPIN); oappend(");"); + oappend(SET_F("xOpt(ux+':digitalmic:pin[]',4,' ⎌',")); oappendi(ES7243_SDAPIN); oappend(");"); #endif - oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',5,'','I2C SCL');")); - oappend(SET_F("rOpt('AudioReactive:digitalmic:pin[]',5,'use global (")); oappendi(i2c_scl); oappend(")',-1);"); + oappend(SET_F("addInfo(ux+':digitalmic:pin[]',5,'','I2C SCL');")); + oappend(SET_F("rOpt(ux+':digitalmic:pin[]',5,'use global (")); oappendi(i2c_scl); oappend(")',-1);"); #ifdef ES7243_SCLPIN - oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',5,' ⎌',")); oappendi(ES7243_SCLPIN); oappend(");"); + oappend(SET_F("xOpt(ux+':digitalmic:pin[]',5,' ⎌',")); oappendi(ES7243_SCLPIN); oappend(");"); #endif - oappend(SET_F("dRO('AudioReactive:digitalmic:pin[]',5);")); // disable read only pins + oappend(SET_F("dRO(ux+':digitalmic:pin[]',5);")); // disable read only pins #endif } From 29013a3f839bf52f378bd50e0e3c3c8829648bf0 Mon Sep 17 00:00:00 2001 From: Frank Date: Mon, 1 Jul 2024 22:54:24 +0200 Subject: [PATCH 17/29] auto-replace long functions with short names adI = addInfo adD = addDropdown adO = addOption adF = addField before: String buffer usage: 3183 of 4037 bytes after: String buffer usage: 2805 of 4037 bytes --- wled00/data/settings_um.htm | 12 ++++++++++++ wled00/fcn_declare.h | 1 + wled00/util.cpp | 18 +++++++++++++++--- 3 files changed, 28 insertions(+), 3 deletions(-) diff --git a/wled00/data/settings_um.htm b/wled00/data/settings_um.htm index 759643e6..19a8590f 100644 --- a/wled00/data/settings_um.htm +++ b/wled00/data/settings_um.htm @@ -146,6 +146,9 @@ urows += `
`; } } + function adF(k,f,o,a=false) { //shortcut for addField(key, field, (sub)object, isArray) + return addField(k,f,o,a); + } // https://stackoverflow.com/questions/39729741/javascript-change-input-text-to-select-option function addDropdown(um,fld) { let sel = d.createElement('select'); @@ -170,6 +173,9 @@ } return null; } + function adD(um,fld) { // shortcut for addDropdown(um,fld) + return addDropdown(um,fld); + } function addOption(sel,txt,val) { if (sel===null) return; // select object missing let opt = d.createElement("option"); @@ -181,6 +187,9 @@ if (c.value == sel.dataset.val) sel.selectedIndex = i; } } + function adO(sel,txt,val) { // shortcut for addOption(sel,txt,val) + return addOption(sel,txt,val); + } //WLEDMM: replace Option to set globals function rOpt(name,el,txt,val) { let obj = d.getElementsByName(name); @@ -285,6 +294,9 @@ if (txt2!="") obj[el].insertAdjacentHTML('beforebegin', txt2 + ' '); //add pre texts } } + function adI(name,el,txt, txt2="") { // shortcut for addInfo(name,el,txt, txt2="") + return addInfo(name,el,txt, txt2); + } // add Help Button function addHB(um) { diff --git a/wled00/fcn_declare.h b/wled00/fcn_declare.h index 38a18b09..aa0645d0 100644 --- a/wled00/fcn_declare.h +++ b/wled00/fcn_declare.h @@ -362,6 +362,7 @@ int getNumVal(const String* req, uint16_t pos); void parseNumber(const char* str, byte* val, byte minv=0, byte maxv=255); bool getVal(JsonVariant elem, byte* val, byte minv=0, byte maxv=255); bool updateVal(const char* req, const char* key, byte* val, byte minv=0, byte maxv=255); +void oappendUseDeflate(bool OnOff); // enable / disable string squeezing bool oappend(const char* txt); // append new c string to temp buffer efficiently bool oappendi(int i); // append new number to temp buffer efficiently void sappend(char stype, const char* key, int val); diff --git a/wled00/util.cpp b/wled00/util.cpp index 896b57e3..1bcf2ad0 100644 --- a/wled00/util.cpp +++ b/wled00/util.cpp @@ -144,22 +144,34 @@ bool oappendi(int i) return oappend(s); } +static bool squeezeStrings = false; +void oappendUseDeflate(bool OnOff) { squeezeStrings = OnOff; } bool oappend(const char* txt) { - uint16_t len = strlen(txt); + String str = squeezeStrings ? String(txt) : String(""); + if (squeezeStrings) { + // simple fixed-dictionary deflate + str.replace(F("addField("), F("adF(")); + str.replace(F("addDropdown("), F("adD(")); + str.replace(F("addOption("), F("adO(")); + str.replace(F("addInfo("), F("adI(")); + } + const char* finalTxt = squeezeStrings ? str.c_str() : txt; + + size_t len = strlen(finalTxt); if ((obuf == nullptr) || (olen + len >= SETTINGS_STACK_BUF_SIZE)) { // sanity checks if (obuf == nullptr) { USER_PRINTLN(F("oappend() error: obuf == nullptr.")); } else { USER_PRINT(F("oappend() error: buffer full. Increase SETTINGS_STACK_BUF_SIZE for ")); USER_PRINTF("%2u bytes \t\"", len /*1 + olen + len - SETTINGS_STACK_BUF_SIZE*/); - USER_PRINT(txt); + USER_PRINT(finalTxt); USER_PRINTLN(F("\"")); errorFlag = ERR_LOW_AJAX_MEM; } return false; // buffer full } - strcpy(obuf + olen, txt); + strcpy(obuf + olen, finalTxt); olen += len; return true; } From 8cfd0df71037d9a073a9a856edb4edd3ffdbffcb Mon Sep 17 00:00:00 2001 From: Frank Date: Mon, 1 Jul 2024 22:54:51 +0200 Subject: [PATCH 18/29] Update xml.cpp --- wled00/xml.cpp | 2 ++ 1 file changed, 2 insertions(+) diff --git a/wled00/xml.cpp b/wled00/xml.cpp index 72fdf219..efc5840c 100644 --- a/wled00/xml.cpp +++ b/wled00/xml.cpp @@ -792,6 +792,7 @@ void getSettingsJS(AsyncWebServerRequest* request, byte subPage, char* dest) //W if (subPage == 8) //usermods { appendGPIOinfo(); + oappendUseDeflate(true); // allow replacing long functions with shorter equivalents - only works for usermods if (!request->hasParam("um") ) { // oappend(SET_F("numM=")); // oappendi(usermods.getModCount()); @@ -834,6 +835,7 @@ void getSettingsJS(AsyncWebServerRequest* request, byte subPage, char* dest) //W Usermod *usermod = usermods.lookupName(request->getParam("um")->value().c_str()); if (usermod) usermod->appendConfigData(); } + oappendUseDeflate(false); // oappend(SET_F("console.log('getSettingsJS fix ro pins', d.max_gpio, d.ro_gpio);")); oappend(SET_F("pinPost();")); From 27dd6bddac5d951637788aba52ac476f9806681c Mon Sep 17 00:00:00 2001 From: Frank Date: Mon, 1 Jul 2024 22:56:18 +0200 Subject: [PATCH 19/29] AudioReactive:digitalmic:pin[] --> uxp before: String buffer usage: 2805 of 4037 bytes after: String buffer usage: 2566 of 4037 bytes --- usermods/audioreactive/audio_reactive.h | 43 +++++++++++++------------ 1 file changed, 22 insertions(+), 21 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 10748b57..10e95153 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -2738,7 +2738,8 @@ class AudioReactive : public Usermod { void appendConfigData() { - oappend(SET_F("ux='AudioReactive';")); // fingers crossed that "ux" isn't already used as JS var, html post parameter or css style + oappend(SET_F("ux='AudioReactive';")); // ux = shortcut for Audioreactive - fingers crossed that "ux" isn't already used as JS var, html post parameter or css style + oappend(SET_F("uxp=ux+':digitalmic:pin[]';")); // uxp = shortcut for AudioReactive:digitalmic:pin[] oappend(SET_F("addInfo(ux+':help',0,'');")); #ifdef ARDUINO_ARCH_ESP32 //WLEDMM: add defaults @@ -2919,45 +2920,45 @@ class AudioReactive : public Usermod { oappend(SET_F("addInfo(ux+':digitalmic:type',1,'requires reboot!');")); // 0 is field type, 1 is actual field #ifdef ARDUINO_ARCH_ESP32 - oappend(SET_F("addInfo(ux+':digitalmic:pin[]',0,'sd/data/dout','I2S SD');")); + oappend(SET_F("addInfo(uxp,0,'sd/data/dout','I2S SD');")); #ifdef I2S_SDPIN - oappend(SET_F("xOpt(ux+':digitalmic:pin[]',0,' ⎌',")); oappendi(I2S_SDPIN); oappend(");"); + oappend(SET_F("xOpt(uxp,0,' ⎌',")); oappendi(I2S_SDPIN); oappend(");"); #endif - oappend(SET_F("addInfo(ux+':digitalmic:pin[]',1,'ws/clk/lrck','I2S WS');")); - oappend(SET_F("dRO(ux+':digitalmic:pin[]',1);")); // disable read only pins + oappend(SET_F("addInfo(uxp,1,'ws/clk/lrck','I2S WS');")); + oappend(SET_F("dRO(uxp,1);")); // disable read only pins #ifdef I2S_WSPIN - oappend(SET_F("xOpt(ux+':digitalmic:pin[]',1,' ⎌',")); oappendi(I2S_WSPIN); oappend(");"); + oappend(SET_F("xOpt(uxp,1,' ⎌',")); oappendi(I2S_WSPIN); oappend(");"); #endif - oappend(SET_F("addInfo(ux+':digitalmic:pin[]',2,'sck/bclk','I2S SCK');")); - oappend(SET_F("dRO(ux+':digitalmic:pin[]',2);")); // disable read only pins + oappend(SET_F("addInfo(uxp,2,'sck/bclk','I2S SCK');")); + oappend(SET_F("dRO(uxp,2);")); // disable read only pins #ifdef I2S_CKPIN - oappend(SET_F("xOpt(ux+':digitalmic:pin[]',2,' ⎌',")); oappendi(I2S_CKPIN); oappend(");"); + oappend(SET_F("xOpt(uxp,2,' ⎌',")); oappendi(I2S_CKPIN); oappend(");"); #endif - oappend(SET_F("addInfo(ux+':digitalmic:pin[]',3,'master clock','I2S MCLK');")); - oappend(SET_F("dRO(ux+':digitalmic:pin[]',3);")); // disable read only pins + oappend(SET_F("addInfo(uxp,3,'master clock','I2S MCLK');")); + oappend(SET_F("dRO(uxp,3);")); // disable read only pins #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) - oappend(SET_F("dOpt(ux+':digitalmic:pin[]',3,2,2);")); //only use -1, 0, 1 or 3 - oappend(SET_F("dOpt(ux+':digitalmic:pin[]',3,4,39);")); //only use -1, 0, 1 or 3 + oappend(SET_F("dOpt(uxp,3,2,2);")); //only use -1, 0, 1 or 3 + oappend(SET_F("dOpt(uxp,3,4,39);")); //only use -1, 0, 1 or 3 #endif #ifdef MCLK_PIN - oappend(SET_F("xOpt(ux+':digitalmic:pin[]',3,' ⎌',")); oappendi(MCLK_PIN); oappend(");"); + oappend(SET_F("xOpt(uxp,3,' ⎌',")); oappendi(MCLK_PIN); oappend(");"); #endif - oappend(SET_F("addInfo(ux+':digitalmic:pin[]',4,'','I2C SDA');")); - oappend(SET_F("rOpt(ux+':digitalmic:pin[]',4,'use global (")); oappendi(i2c_sda); oappend(")',-1);"); + oappend(SET_F("addInfo(uxp,4,'','I2C SDA');")); + oappend(SET_F("rOpt(uxp,4,'use global (")); oappendi(i2c_sda); oappend(")',-1);"); #ifdef ES7243_SDAPIN - oappend(SET_F("xOpt(ux+':digitalmic:pin[]',4,' ⎌',")); oappendi(ES7243_SDAPIN); oappend(");"); + oappend(SET_F("xOpt(uxp,4,' ⎌',")); oappendi(ES7243_SDAPIN); oappend(");"); #endif - oappend(SET_F("addInfo(ux+':digitalmic:pin[]',5,'','I2C SCL');")); - oappend(SET_F("rOpt(ux+':digitalmic:pin[]',5,'use global (")); oappendi(i2c_scl); oappend(")',-1);"); + oappend(SET_F("addInfo(uxp,5,'','I2C SCL');")); + oappend(SET_F("rOpt(uxp,5,'use global (")); oappendi(i2c_scl); oappend(")',-1);"); #ifdef ES7243_SCLPIN - oappend(SET_F("xOpt(ux+':digitalmic:pin[]',5,' ⎌',")); oappendi(ES7243_SCLPIN); oappend(");"); + oappend(SET_F("xOpt(uxp,5,' ⎌',")); oappendi(ES7243_SCLPIN); oappend(");"); #endif - oappend(SET_F("dRO(ux+':digitalmic:pin[]',5);")); // disable read only pins + oappend(SET_F("dRO(uxp,5);")); // disable read only pins #endif } From 788347440466e84df4c58d899aa1081c6b3251b3 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Thu, 4 Jul 2024 18:37:31 +0200 Subject: [PATCH 20/29] postProcessFFTResults minor refactoring * limiter on/off : made the logic a bit clearer. * use a simpler way of writing filters: a = n*a + (1-n)*b --> a = a+ n*(b-a) no functional impact. --- usermods/audioreactive/audio_reactive.h | 42 ++++++++++++++----------- 1 file changed, 24 insertions(+), 18 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index b6983c6d..79172591 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -946,27 +946,33 @@ static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels) // p if(fftCalc[i] < 0) fftCalc[i] = 0; } - // smooth results - rise fast, fall slower - if(fftCalc[i] > fftAvg[i]) // rise fast - fftAvg[i] = fftCalc[i] *0.78f + 0.22f*fftAvg[i]; // will need approx 1-2 cycles (50ms) for converging against fftCalc[i] - else { // fall slow - if (decayTime < 250) fftAvg[i] = fftCalc[i]*0.4f + 0.6f*fftAvg[i]; - else if (decayTime < 500) fftAvg[i] = fftCalc[i]*0.33f + 0.67f*fftAvg[i]; - else if (decayTime < 1000) fftAvg[i] = fftCalc[i]*0.22f + 0.78f*fftAvg[i]; // approx 5 cycles (225ms) for falling to zero - else if (decayTime < 2000) fftAvg[i] = fftCalc[i]*0.17f + 0.83f*fftAvg[i]; // default - approx 9 cycles (225ms) for falling to zero - else if (decayTime < 3000) fftAvg[i] = fftCalc[i]*0.14f + 0.86f*fftAvg[i]; // approx 14 cycles (350ms) for falling to zero - else if (decayTime < 4000) fftAvg[i] = fftCalc[i]*0.1f + 0.9f*fftAvg[i]; - else fftAvg[i] = fftCalc[i]*0.05f + 0.95f*fftAvg[i]; - } - // constrain internal vars - just to be sure - fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f); - fftAvg[i] = constrain(fftAvg[i], 0.0f, 1023.0f); - float currentResult; - if(limiterOn == true) + if(limiterOn == true) { + // Limiter ON -> smooth results -> rise fast, fall slower + if(fftCalc[i] > fftAvg[i]) { // rise fast + fftAvg[i] += 0.78f * (fftCalc[i] - fftAvg[i]); // will need approx 1-2 cycles (50ms) for converging against fftCalc[i] + } else { // fall slow + if (decayTime < 150) fftAvg[i] += 0.50f * (fftCalc[i] - fftAvg[i]); + else if (decayTime < 250) fftAvg[i] += 0.40f * (fftCalc[i] - fftAvg[i]); + else if (decayTime < 500) fftAvg[i] += 0.33f * (fftCalc[i] - fftAvg[i]); + else if (decayTime < 1000) fftAvg[i] += 0.22f * (fftCalc[i] - fftAvg[i]); // approx 5 cycles (225ms) for falling to zero + else if (decayTime < 2000) fftAvg[i] += 0.17f * (fftCalc[i] - fftAvg[i]); // default - approx 9 cycles (225ms) for falling to zero + else if (decayTime < 3000) fftAvg[i] += 0.14f * (fftCalc[i] - fftAvg[i]); // approx 14 cycles (350ms) for falling to zero + else if (decayTime < 4000) fftAvg[i] += 0.10f * (fftCalc[i] - fftAvg[i]); + else fftAvg[i] += 0.05f * (fftCalc[i] - fftAvg[i]); + } + // constrain internal vars - just to be sure + fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f); + fftAvg[i] = constrain(fftAvg[i], 0.0f, 1023.0f); + // use filtered result currentResult = fftAvg[i]; - else + } else { + // Limiter OFF -> no adjustments + fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f); + fftAvg[i] = fftCalc[i]; // keep filters up-to-date + // use not filtered result currentResult = fftCalc[i]; + } switch (FFTScalingMode) { case 1: From 7d04660b913fedecc62f635f519444866e2ac4a2 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Thu, 4 Jul 2024 20:06:13 +0200 Subject: [PATCH 21/29] postProcessFFTResults adjustments for FastPath samples arrive 2x as fast in fast mode - adjust filters accordingly --- usermods/audioreactive/audio_reactive.h | 58 ++++++++++++++----------- 1 file changed, 33 insertions(+), 25 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 79172591..66175681 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -285,7 +285,7 @@ static float mapf(float x, float in_min, float in_max, float out_min, float out_ static float fftAddAvg(int from, int to); // average of several FFT result bins void FFTcode(void * parameter); // audio processing task: read samples, run FFT, fill GEQ channels from FFT results static void runMicFilter(uint16_t numSamples, float *sampleBuffer); // pre-filtering of raw samples (band-pass) -static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels); // post-processing and post-amp of GEQ channels +static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels, bool i2sFastpath); // post-processing and post-amp of GEQ channels static TaskHandle_t FFT_Task = nullptr; @@ -860,8 +860,8 @@ void FFTcode(void * parameter) // post-processing of frequency channels (pink noise adjustment, AGC, smoothing, scaling) if (pinkIndex > MAX_PINK) pinkIndex = MAX_PINK; - //postProcessFFTResults((fabsf(sampleAvg) > 0.25f)? true : false , NUM_GEQ_CHANNELS); - postProcessFFTResults((fabsf(volumeSmth)>0.25f)? true : false , NUM_GEQ_CHANNELS); // this function modifies fftCalc, fftAvg and fftResult + + postProcessFFTResults((fabsf(volumeSmth) > 0.25f)? true : false, NUM_GEQ_CHANNELS, usingOldSamples); // this function modifies fftCalc, fftAvg and fftResult #if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS) // timing @@ -933,7 +933,7 @@ static void runMicFilter(uint16_t numSamples, float *sampleBuffer) // p } } -static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels) // post-processing and post-amp of GEQ channels +static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels, bool i2sFastpath) // post-processing and post-amp of GEQ channels { for (int i=0; i < numberOfChannels; i++) { @@ -946,34 +946,42 @@ static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels) // p if(fftCalc[i] < 0) fftCalc[i] = 0; } - float currentResult; + float speed = 1.0f; // filter correction for sampling speed -> 1.0 in normal mode (43hz) + if (i2sFastpath) speed = 0.6931471805599453094f * 1.1f; // -> ln(2) from math, *1.1 from my gut feeling ;-) in fast mode (86hz) + if(limiterOn == true) { - // Limiter ON -> smooth results -> rise fast, fall slower + // Limiter ON -> smooth results if(fftCalc[i] > fftAvg[i]) { // rise fast - fftAvg[i] += 0.78f * (fftCalc[i] - fftAvg[i]); // will need approx 1-2 cycles (50ms) for converging against fftCalc[i] + fftAvg[i] += speed * 0.78f * (fftCalc[i] - fftAvg[i]); // will need approx 1-2 cycles (50ms) for converging against fftCalc[i] } else { // fall slow - if (decayTime < 150) fftAvg[i] += 0.50f * (fftCalc[i] - fftAvg[i]); - else if (decayTime < 250) fftAvg[i] += 0.40f * (fftCalc[i] - fftAvg[i]); - else if (decayTime < 500) fftAvg[i] += 0.33f * (fftCalc[i] - fftAvg[i]); - else if (decayTime < 1000) fftAvg[i] += 0.22f * (fftCalc[i] - fftAvg[i]); // approx 5 cycles (225ms) for falling to zero - else if (decayTime < 2000) fftAvg[i] += 0.17f * (fftCalc[i] - fftAvg[i]); // default - approx 9 cycles (225ms) for falling to zero - else if (decayTime < 3000) fftAvg[i] += 0.14f * (fftCalc[i] - fftAvg[i]); // approx 14 cycles (350ms) for falling to zero - else if (decayTime < 4000) fftAvg[i] += 0.10f * (fftCalc[i] - fftAvg[i]); - else fftAvg[i] += 0.05f * (fftCalc[i] - fftAvg[i]); + if (decayTime < 150) fftAvg[i] += speed * 0.50f * (fftCalc[i] - fftAvg[i]); + else if (decayTime < 250) fftAvg[i] += speed * 0.40f * (fftCalc[i] - fftAvg[i]); + else if (decayTime < 500) fftAvg[i] += speed * 0.33f * (fftCalc[i] - fftAvg[i]); + else if (decayTime < 1000) fftAvg[i] += speed * 0.22f * (fftCalc[i] - fftAvg[i]); // approx 5 cycles (225ms) for falling to zero + else if (decayTime < 2000) fftAvg[i] += speed * 0.17f * (fftCalc[i] - fftAvg[i]); // default - approx 9 cycles (225ms) for falling to zero + else if (decayTime < 3000) fftAvg[i] += speed * 0.14f * (fftCalc[i] - fftAvg[i]); // approx 14 cycles (350ms) for falling to zero + else if (decayTime < 4000) fftAvg[i] += speed * 0.10f * (fftCalc[i] - fftAvg[i]); + else fftAvg[i] += speed * 0.05f * (fftCalc[i] - fftAvg[i]); } - // constrain internal vars - just to be sure - fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f); - fftAvg[i] = constrain(fftAvg[i], 0.0f, 1023.0f); - // use filtered result - currentResult = fftAvg[i]; } else { - // Limiter OFF -> no adjustments - fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f); - fftAvg[i] = fftCalc[i]; // keep filters up-to-date - // use not filtered result - currentResult = fftCalc[i]; + // Limiter OFF + if (i2sFastpath) { + // fast mode -> average last two results + float tmp = fftCalc[i]; + fftCalc[i] = 0.7f * tmp + 0.3f * fftAvg[i]; + fftAvg[i] = tmp; // store current sample for next run + } else { + // normal mode -> no adjustments + fftAvg[i] = fftCalc[i]; // keep filters up-to-date + } } + // constrain internal vars - just to be sure + fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f); + fftAvg[i] = constrain(fftAvg[i], 0.0f, 1023.0f); + + float currentResult = limiterOn ? fftAvg[i] : fftCalc[i]; // continue with filtered result (limiter on) or unfiltered result (limiter off) + switch (FFTScalingMode) { case 1: // Logarithmic scaling From 9b14de06b2b1229c9f43bd3e3134691a1b687498 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Thu, 4 Jul 2024 20:13:57 +0200 Subject: [PATCH 22/29] fix build error on -S2, -C3 --- usermods/audioreactive/audio_reactive.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 66175681..8f4ed3af 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -861,7 +861,11 @@ void FFTcode(void * parameter) // post-processing of frequency channels (pink noise adjustment, AGC, smoothing, scaling) if (pinkIndex > MAX_PINK) pinkIndex = MAX_PINK; +#ifdef FFT_USE_SLIDING_WINDOW postProcessFFTResults((fabsf(volumeSmth) > 0.25f)? true : false, NUM_GEQ_CHANNELS, usingOldSamples); // this function modifies fftCalc, fftAvg and fftResult +#else + postProcessFFTResults((fabsf(volumeSmth) > 0.25f)? true : false, NUM_GEQ_CHANNELS, false); // this function modifies fftCalc, fftAvg and fftResult +#endif #if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS) // timing From 160e66a766b2905d4fb496fc108e178f3542624d Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Thu, 4 Jul 2024 21:44:09 +0200 Subject: [PATCH 23/29] user-selectable FFT window functions experiments: user-selectable FFT "windowing" options https://en.wikipedia.org/wiki/Window_function#Cosine-sum_windows --- usermods/audioreactive/audio_reactive.h | 164 ++++++++++++++---------- 1 file changed, 99 insertions(+), 65 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 8f4ed3af..255815fd 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -179,7 +179,7 @@ static bool limiterOn = false; // bool: enable / disable dynamic static bool limiterOn = true; #endif #ifdef FFT_USE_SLIDING_WINDOW -static uint16_t attackTime = 14; // int: attack time in milliseconds. Default 0.014sec +static uint16_t attackTime = 24; // int: attack time in milliseconds. Default 0.024sec static uint16_t decayTime = 250; // int: decay time in milliseconds. New default 250ms. #else static uint16_t attackTime = 50; // int: attack time in milliseconds. Default 0.08sec @@ -257,6 +257,7 @@ static constexpr uint8_t averageByRMS = false; // false: us static constexpr uint8_t averageByRMS = true; // false: use mean value, true: use RMS (root mean squared). use better method on fast MCUs. #endif static uint8_t freqDist = 0; // 0=old 1=rightshift mode +static uint8_t fftWindow = 0; // FFT windowing function (0 = default) #ifdef FFT_USE_SLIDING_WINDOW static uint8_t doSlidingFFT = 1; // 1 = use sliding window FFT (faster & more accurate) #endif @@ -685,17 +686,41 @@ void FFTcode(void * parameter) micReal_max2 = datMax; #endif #endif + + float wc = 1.0; // FFT window correction factor, relative to Blackman_Harris + // run FFT (takes 3-5ms on ESP32) //if (fabsf(sampleAvg) > 0.25f) { // noise gate open if (fabsf(volumeSmth) > 0.25f) { // noise gate open if ((skipSecondFFT == false) || (isFirstRun == true)) { // run FFT (takes 2-3ms on ESP32, ~12ms on ESP32-S2, ~30ms on -C3) if (doDCRemoval) FFT.dcRemoval(); // remove DC offset - #if !defined(FFT_PREFER_EXACT_PEAKS) - FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude accuracy - #else - FFT.windowing(FFTWindow::Blackman_Harris, FFTDirection::Forward); // Weigh data using "Blackman- Harris" window - sharp peaks due to excellent sideband rejection + switch(fftWindow) { // apply FFT window + case 1: + FFT.windowing(FFTWindow::Hann, FFTDirection::Forward); // recommended for 50% overlap + wc = 0.66415918066; // 1.8554726898 * 2.0 + break; + case 2: + FFT.windowing( FFTWindow::Nuttall, FFTDirection::Forward); + wc = 0.9916873881f; // 2.8163172034 * 2.0 + break; + case 3: + FFT.windowing( FFTWindow::Hamming, FFTDirection::Forward); + wc = 0.664159180663f; // 1.8549343278 * 2.0 + break; + case 4: + FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude preservation, low frequency accuracy + wc = 1.276771793156f; // 3.5659039231 * 2.0 + break; + case 0: // falls through + default: + FFT.windowing(FFTWindow::Blackman_Harris, FFTDirection::Forward); // Weigh data using "Blackman- Harris" window - sharp peaks due to excellent sideband rejection + wc = 1.0f; // 2.7929062517 * 2.0 + } + #ifdef FFT_USE_SLIDING_WINDOW + if (usingOldSamples) wc = wc * 1.10f; // compensate for loss caused by averaging #endif + FFT.compute( FFTDirection::Forward ); // Compute FFT FFT.complexToMagnitude(); // Compute magnitudes vReal[0] = 0; // The remaining DC offset on the signal produces a strong spike on position 0 that should be eliminated to avoid issues. @@ -715,6 +740,7 @@ void FFTcode(void * parameter) #else FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant #endif + FFT_Magnitude *= wc; // apply correction factor if (FFT_MajorPeak < (SAMPLE_RATE / samplesFFT)) {FFT_MajorPeak = 1.0f; FFT_Magnitude = 0;} // too low - use zero if (FFT_MajorPeak > (0.42f * SAMPLE_RATE)) {FFT_MajorPeak = last_majorpeak; FFT_Magnitude = last_magnitude;} // too high - keep last peak @@ -764,23 +790,23 @@ void FFTcode(void * parameter) * End frequency = Start frequency * multiplier ^ 16 * Multiplier = (End frequency/ Start frequency) ^ 1/16 * Multiplier = 1.320367784 - */ // Range - fftCalc[ 0] = fftAddAvg(2,4); // 60 - 100 - fftCalc[ 1] = fftAddAvg(4,5); // 80 - 120 - fftCalc[ 2] = fftAddAvg(5,7); // 100 - 160 - fftCalc[ 3] = fftAddAvg(7,9); // 140 - 200 - fftCalc[ 4] = fftAddAvg(9,12); // 180 - 260 - fftCalc[ 5] = fftAddAvg(12,16); // 240 - 340 - fftCalc[ 6] = fftAddAvg(16,21); // 320 - 440 - fftCalc[ 7] = fftAddAvg(21,29); // 420 - 600 - fftCalc[ 8] = fftAddAvg(29,37); // 580 - 760 - fftCalc[ 9] = fftAddAvg(37,48); // 740 - 980 - fftCalc[10] = fftAddAvg(48,64); // 960 - 1300 - fftCalc[11] = fftAddAvg(64,84); // 1280 - 1700 - fftCalc[12] = fftAddAvg(84,111); // 1680 - 2240 - fftCalc[13] = fftAddAvg(111,147); // 2220 - 2960 - fftCalc[14] = fftAddAvg(147,194); // 2940 - 3900 - fftCalc[15] = fftAddAvg(194,250); // 3880 - 5000 // avoid the last 5 bins, which are usually inaccurate + */ // Range + fftCalc[ 0] = wc * fftAddAvg(2,4); // 60 - 100 + fftCalc[ 1] = wc * fftAddAvg(4,5); // 80 - 120 + fftCalc[ 2] = wc * fftAddAvg(5,7); // 100 - 160 + fftCalc[ 3] = wc * fftAddAvg(7,9); // 140 - 200 + fftCalc[ 4] = wc * fftAddAvg(9,12); // 180 - 260 + fftCalc[ 5] = wc * fftAddAvg(12,16); // 240 - 340 + fftCalc[ 6] = wc * fftAddAvg(16,21); // 320 - 440 + fftCalc[ 7] = wc * fftAddAvg(21,29); // 420 - 600 + fftCalc[ 8] = wc * fftAddAvg(29,37); // 580 - 760 + fftCalc[ 9] = wc * fftAddAvg(37,48); // 740 - 980 + fftCalc[10] = wc * fftAddAvg(48,64); // 960 - 1300 + fftCalc[11] = wc * fftAddAvg(64,84); // 1280 - 1700 + fftCalc[12] = wc * fftAddAvg(84,111); // 1680 - 2240 + fftCalc[13] = wc * fftAddAvg(111,147); // 2220 - 2960 + fftCalc[14] = wc * fftAddAvg(147,194); // 2940 - 3900 + fftCalc[15] = wc * fftAddAvg(194,250); // 3880 - 5000 // avoid the last 5 bins, which are usually inaccurate #else //WLEDMM: different distributions if (freqDist == 0) { @@ -788,60 +814,60 @@ void FFTcode(void * parameter) // bins frequency range if (useInputFilter==1) { // skip frequencies below 100hz - fftCalc[ 0] = 0.8f * fftAddAvg(3,3); - fftCalc[ 1] = 0.9f * fftAddAvg(4,4); - fftCalc[ 2] = fftAddAvg(5,5); - fftCalc[ 3] = fftAddAvg(6,6); + fftCalc[ 0] = wc * 0.8f * fftAddAvg(3,3); + fftCalc[ 1] = wc * 0.9f * fftAddAvg(4,4); + fftCalc[ 2] = wc * fftAddAvg(5,5); + fftCalc[ 3] = wc * fftAddAvg(6,6); // don't use the last bins from 206 to 255. - fftCalc[15] = fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping + fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping } else { - fftCalc[ 0] = fftAddAvg(1,1); // 1 43 - 86 sub-bass - fftCalc[ 1] = fftAddAvg(2,2); // 1 86 - 129 bass - fftCalc[ 2] = fftAddAvg(3,4); // 2 129 - 216 bass - fftCalc[ 3] = fftAddAvg(5,6); // 2 216 - 301 bass + midrange + fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass + fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass + fftCalc[ 2] = wc * fftAddAvg(3,4); // 2 129 - 216 bass + fftCalc[ 3] = wc * fftAddAvg(5,6); // 2 216 - 301 bass + midrange // don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise) - fftCalc[15] = fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping + fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping } - fftCalc[ 4] = fftAddAvg(7,9); // 3 301 - 430 midrange - fftCalc[ 5] = fftAddAvg(10,12); // 3 430 - 560 midrange - fftCalc[ 6] = fftAddAvg(13,18); // 5 560 - 818 midrange - fftCalc[ 7] = fftAddAvg(19,25); // 7 818 - 1120 midrange -- 1Khz should always be the center ! - fftCalc[ 8] = fftAddAvg(26,32); // 7 1120 - 1421 midrange - fftCalc[ 9] = fftAddAvg(33,43); // 9 1421 - 1895 midrange - fftCalc[10] = fftAddAvg(44,55); // 12 1895 - 2412 midrange + high mid - fftCalc[11] = fftAddAvg(56,69); // 14 2412 - 3015 high mid - fftCalc[12] = fftAddAvg(70,85); // 16 3015 - 3704 high mid - fftCalc[13] = fftAddAvg(86,103); // 18 3704 - 4479 high mid - fftCalc[14] = fftAddAvg(104,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping + fftCalc[ 4] = wc * fftAddAvg(7,9); // 3 301 - 430 midrange + fftCalc[ 5] = wc * fftAddAvg(10,12); // 3 430 - 560 midrange + fftCalc[ 6] = wc * fftAddAvg(13,18); // 5 560 - 818 midrange + fftCalc[ 7] = wc * fftAddAvg(19,25); // 7 818 - 1120 midrange -- 1Khz should always be the center ! + fftCalc[ 8] = wc * fftAddAvg(26,32); // 7 1120 - 1421 midrange + fftCalc[ 9] = wc * fftAddAvg(33,43); // 9 1421 - 1895 midrange + fftCalc[10] = wc * fftAddAvg(44,55); // 12 1895 - 2412 midrange + high mid + fftCalc[11] = wc * fftAddAvg(56,69); // 14 2412 - 3015 high mid + fftCalc[12] = wc * fftAddAvg(70,85); // 16 3015 - 3704 high mid + fftCalc[13] = wc * fftAddAvg(86,103); // 18 3704 - 4479 high mid + fftCalc[14] = wc * fftAddAvg(104,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping } else if (freqDist == 1) { //WLEDMM: Rightshift: note ewowi: frequencies in comments are not correct if (useInputFilter==1) { // skip frequencies below 100hz - fftCalc[ 0] = 0.8f * fftAddAvg(1,1); - fftCalc[ 1] = 0.9f * fftAddAvg(2,2); - fftCalc[ 2] = fftAddAvg(3,3); - fftCalc[ 3] = fftAddAvg(4,4); + fftCalc[ 0] = wc * 0.8f * fftAddAvg(1,1); + fftCalc[ 1] = wc * 0.9f * fftAddAvg(2,2); + fftCalc[ 2] = wc * fftAddAvg(3,3); + fftCalc[ 3] = wc * fftAddAvg(4,4); // don't use the last bins from 206 to 255. - fftCalc[15] = fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping + fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping } else { - fftCalc[ 0] = fftAddAvg(1,1); // 1 43 - 86 sub-bass - fftCalc[ 1] = fftAddAvg(2,2); // 1 86 - 129 bass - fftCalc[ 2] = fftAddAvg(3,3); // 2 129 - 216 bass - fftCalc[ 3] = fftAddAvg(4,4); // 2 216 - 301 bass + midrange + fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass + fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass + fftCalc[ 2] = wc * fftAddAvg(3,3); // 2 129 - 216 bass + fftCalc[ 3] = wc * fftAddAvg(4,4); // 2 216 - 301 bass + midrange // don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise) - fftCalc[15] = fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping + fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping } - fftCalc[ 4] = fftAddAvg(5,6); // 3 301 - 430 midrange - fftCalc[ 5] = fftAddAvg(7,8); // 3 430 - 560 midrange - fftCalc[ 6] = fftAddAvg(9,10); // 5 560 - 818 midrange - fftCalc[ 7] = fftAddAvg(11,13); // 7 818 - 1120 midrange -- 1Khz should always be the center ! - fftCalc[ 8] = fftAddAvg(14,18); // 7 1120 - 1421 midrange - fftCalc[ 9] = fftAddAvg(19,25); // 9 1421 - 1895 midrange - fftCalc[10] = fftAddAvg(26,36); // 12 1895 - 2412 midrange + high mid - fftCalc[11] = fftAddAvg(37,45); // 14 2412 - 3015 high mid - fftCalc[12] = fftAddAvg(46,66); // 16 3015 - 3704 high mid - fftCalc[13] = fftAddAvg(67,97); // 18 3704 - 4479 high mid - fftCalc[14] = fftAddAvg(98,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping + fftCalc[ 4] = wc * fftAddAvg(5,6); // 3 301 - 430 midrange + fftCalc[ 5] = wc * fftAddAvg(7,8); // 3 430 - 560 midrange + fftCalc[ 6] = wc * fftAddAvg(9,10); // 5 560 - 818 midrange + fftCalc[ 7] = wc * fftAddAvg(11,13); // 7 818 - 1120 midrange -- 1Khz should always be the center ! + fftCalc[ 8] = wc * fftAddAvg(14,18); // 7 1120 - 1421 midrange + fftCalc[ 9] = wc * fftAddAvg(19,25); // 9 1421 - 1895 midrange + fftCalc[10] = wc * fftAddAvg(26,36); // 12 1895 - 2412 midrange + high mid + fftCalc[11] = wc * fftAddAvg(37,45); // 14 2412 - 3015 high mid + fftCalc[12] = wc * fftAddAvg(46,66); // 16 3015 - 3704 high mid + fftCalc[13] = wc * fftAddAvg(67,97); // 18 3704 - 4479 high mid + fftCalc[14] = wc * fftAddAvg(98,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping } #endif } else { // noise gate closed - just decay old values @@ -2756,7 +2782,7 @@ class AudioReactive : public Usermod { poweruser[F("micLev")] = micLevelMethod; poweruser[F("freqDist")] = freqDist; //poweruser[F("freqRMS")] = averageByRMS; - + poweruser[F("FFT_Window")] = fftWindow; #ifdef FFT_USE_SLIDING_WINDOW poweruser[F("I2S_FastPath")] = doSlidingFFT; #endif @@ -2831,6 +2857,7 @@ class AudioReactive : public Usermod { configComplete &= getJsonValue(top["experiments"][F("micLev")], micLevelMethod); configComplete &= getJsonValue(top["experiments"][F("freqDist")], freqDist); //configComplete &= getJsonValue(top["experiments"][F("freqRMS")], averageByRMS); + configComplete &= getJsonValue(top["experiments"][F("FFT_Window")], fftWindow); #ifdef FFT_USE_SLIDING_WINDOW configComplete &= getJsonValue(top["experiments"][F("I2S_FastPath")], doSlidingFFT); #endif @@ -2944,6 +2971,13 @@ class AudioReactive : public Usermod { //oappend(SET_F("addOption(dd,'On',1);")); //oappend(SET_F("addInfo(ux+':experiments:freqRMS',1,'☾');")); + oappend(SET_F("dd=addDropdown(ux,'experiments:FFT_Window');")); + oappend(SET_F("addOption(dd,'Blackman-Harris (MM standard)',0);")); + oappend(SET_F("addOption(dd,'Hann (balanced)',1);")); + oappend(SET_F("addOption(dd,'Nuttall (more accurate)',2);")); + oappend(SET_F("addOption(dd,'Hamming',3);")); + oappend(SET_F("addOption(dd,'Flat-Top (AC WLED, inaccurate)',4);")); + #ifdef FFT_USE_SLIDING_WINDOW oappend(SET_F("dd=addDropdown(ux,'experiments:I2S_FastPath');")); oappend(SET_F("addOption(dd,'Off',0);")); From cba883c663143b2068b0c2c9bca72edb50707635 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Fri, 5 Jul 2024 15:12:41 +0200 Subject: [PATCH 24/29] parameter tuning *some filter parameter tinkering * restore FFT_MajPeakSmth in UDP SoundSync receiver --- usermods/audioreactive/audio_reactive.h | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 255815fd..496f6866 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -240,7 +240,7 @@ const double agcFollowSlow[AGC_NUM_PRESETS] = {1/6144.f,1/4096.f,1/8192.f}; // const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter const double agcControlKi[AGC_NUM_PRESETS] = { 1.7f, 1.85f, 1.2f}; // AGC - PI control, integral gain parameter #if defined(WLEDMM_FASTPATH) -const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/6.f, 1/5.f, 1/10.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value) +const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/8.f, 1/5.f, 1/12.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value) #else const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value) #endif @@ -1434,13 +1434,8 @@ class AudioReactive : public Usermod { { float sampleAdj; // Gain adjusted sample value float tmpSample; // An interim sample variable used for calculations. -#ifdef WLEDMM_FASTPATH - constexpr float weighting = 0.35f; // slightly reduced filter strength, to reduce audio latency - constexpr float weighting2 = 0.25f; -#else - const float weighting = 0.2f; // Exponential filter weighting. Will be adjustable in a future release. + const float weighting = 0.18f; // Exponential filter weighting. Will be adjustable in a future release. const float weighting2 = 0.073f; // Exponential filter weighting, for rising signal (a bit more robust against spikes) -#endif 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 static bool isFrozen = false; static bool haveSilence = true; @@ -1492,9 +1487,11 @@ class AudioReactive : public Usermod { if ((micLevelMethod == 2) && !haveSilence && (expAdjF >= (1.5f * float(soundSquelch)))) isFrozen = true; // fast freeze mode: freeze micLevel once the volume rises 50% above squelch - //expAdjF = (micInNoDC <= soundSquelch) ? 0: expAdjF; // simple noise gate - experimental - expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate - if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0" + // simple noise gate + if ((expAdjF <= soundSquelch) || ((soundSquelch == 0) && (expAdjF < 0.25f))) { + expAdjF = 0.0f; + micInNoDC = 0.0f; + } if (expAdjF <= 0.5f) haveSilence = true; @@ -1515,7 +1512,7 @@ class AudioReactive : public Usermod { sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // Adjust the gain. with inputLevel adjustment sampleReal = tmpSample; - sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ??? + sampleAdj = fmax(fmin(sampleAdj, 255.0f), 0.0f); // Question: why are we limiting the value to 8 bits ??? sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!! // keep "peak" sample, but decay value if current sample is below peak @@ -1538,7 +1535,7 @@ class AudioReactive : public Usermod { if (sampleMax < 0.5f) sampleMax = 0.0f; #if defined(WLEDMM_FASTPATH) - sampleAvg = ((sampleAvg * 7.0f) + sampleAdj) / 8.0f; // make reactions a bit more "crisp" in fastpath mode + sampleAvg = ((sampleAvg * 11.0f) + sampleAdj) / 12.0f; // make reactions a bit more "crisp" in fastpath mode #else sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples. #endif @@ -1791,6 +1788,9 @@ class AudioReactive : public Usermod { my_magnitude = fmaxf(receivedPacket.FFT_Magnitude, 0.0f); FFT_Magnitude = my_magnitude; FFT_MajorPeak = constrain(receivedPacket.FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects +#ifdef ARDUINO_ARCH_ESP32 + FFT_MajPeakSmth = FFT_MajPeakSmth + 0.42f * (FFT_MajorPeak - FFT_MajPeakSmth); // simulate smooth value +#endif agcSensitivity = 128.0f; // substitute - V2 format does not include this value zeroCrossingCount = receivedPacket.zeroCrossingCount; From bd997f7056d0940a77ea325ba5ea185cfd402f89 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Fri, 5 Jul 2024 15:37:03 +0200 Subject: [PATCH 25/29] added Blackman window --- usermods/audioreactive/audio_reactive.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 496f6866..6ec8fb23 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -704,6 +704,10 @@ void FFTcode(void * parameter) FFT.windowing( FFTWindow::Nuttall, FFTDirection::Forward); wc = 0.9916873881f; // 2.8163172034 * 2.0 break; + case 5: + FFT.windowing( FFTWindow::Blackman, FFTDirection::Forward); + wc = 0.84762867875f; // 2.3673474360 * 2.0 + break; case 3: FFT.windowing( FFTWindow::Hamming, FFTDirection::Forward); wc = 0.664159180663f; // 1.8549343278 * 2.0 @@ -2975,6 +2979,7 @@ class AudioReactive : public Usermod { oappend(SET_F("addOption(dd,'Blackman-Harris (MM standard)',0);")); oappend(SET_F("addOption(dd,'Hann (balanced)',1);")); oappend(SET_F("addOption(dd,'Nuttall (more accurate)',2);")); + oappend(SET_F("addOption(dd,'Blackman',5);")); oappend(SET_F("addOption(dd,'Hamming',3);")); oappend(SET_F("addOption(dd,'Flat-Top (AC WLED, inaccurate)',4);")); From 4d03af6466f68891fe1ec696188542c51aac4c13 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Fri, 5 Jul 2024 17:53:43 +0200 Subject: [PATCH 26/29] mic quality dropdown, cleanup * mic quality: When set to "low noise" or "perfect", only minimal smoothing is performed on the "Mic Volume" input. Every filter adds a delay, so this option can lead to better on-spot responses from effects. * cleanup: removed a few unused variables and unused code. --- usermods/audioreactive/audio_reactive.h | 236 +++++++++++------------- 1 file changed, 104 insertions(+), 132 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 6ec8fb23..2de84087 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -178,6 +178,7 @@ static bool limiterOn = false; // bool: enable / disable dynamic #else static bool limiterOn = true; #endif +static uint8_t micQuality = 0; // affects input filtering; 0 normal, 1 minimal filtering, 2 no filtering #ifdef FFT_USE_SLIDING_WINDOW static uint16_t attackTime = 24; // int: attack time in milliseconds. Default 0.024sec static uint16_t decayTime = 250; // int: decay time in milliseconds. New default 250ms. @@ -690,7 +691,6 @@ void FFTcode(void * parameter) float wc = 1.0; // FFT window correction factor, relative to Blackman_Harris // run FFT (takes 3-5ms on ESP32) - //if (fabsf(sampleAvg) > 0.25f) { // noise gate open if (fabsf(volumeSmth) > 0.25f) { // noise gate open if ((skipSecondFFT == false) || (isFirstRun == true)) { // run FFT (takes 2-3ms on ESP32, ~12ms on ESP32-S2, ~30ms on -C3) @@ -776,7 +776,6 @@ void FFTcode(void * parameter) } if ((skipSecondFFT == false) || (isFirstRun == true)) { - 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 vReal[i] = t / 16.0f; // Reduce magnitude. Want end result to be scaled linear and ~4096 max. @@ -785,102 +784,72 @@ void FFTcode(void * parameter) // mapping of FFT result bins to frequency channels //if (fabsf(sampleAvg) > 0.25f) { // noise gate open if (fabsf(volumeSmth) > 0.25f) { // noise gate open -#if 0 - /* This FFT post processing is a DIY endeavour. What we really need is someone with sound engineering expertise to do a great job here AND most importantly, that the animations look GREAT as a result. - * - * Andrew's updated mapping of 256 bins down to the 16 result bins with Sample Freq = 10240, samplesFFT = 512 and some overlap. - * Based on testing, the lowest/Start frequency is 60 Hz (with bin 3) and a highest/End frequency of 5120 Hz in bin 255. - * Now, Take the 60Hz and multiply by 1.320367784 to get the next frequency and so on until the end. Then determine the bins. - * End frequency = Start frequency * multiplier ^ 16 - * Multiplier = (End frequency/ Start frequency) ^ 1/16 - * Multiplier = 1.320367784 - */ // Range - fftCalc[ 0] = wc * fftAddAvg(2,4); // 60 - 100 - fftCalc[ 1] = wc * fftAddAvg(4,5); // 80 - 120 - fftCalc[ 2] = wc * fftAddAvg(5,7); // 100 - 160 - fftCalc[ 3] = wc * fftAddAvg(7,9); // 140 - 200 - fftCalc[ 4] = wc * fftAddAvg(9,12); // 180 - 260 - fftCalc[ 5] = wc * fftAddAvg(12,16); // 240 - 340 - fftCalc[ 6] = wc * fftAddAvg(16,21); // 320 - 440 - fftCalc[ 7] = wc * fftAddAvg(21,29); // 420 - 600 - fftCalc[ 8] = wc * fftAddAvg(29,37); // 580 - 760 - fftCalc[ 9] = wc * fftAddAvg(37,48); // 740 - 980 - fftCalc[10] = wc * fftAddAvg(48,64); // 960 - 1300 - fftCalc[11] = wc * fftAddAvg(64,84); // 1280 - 1700 - fftCalc[12] = wc * fftAddAvg(84,111); // 1680 - 2240 - fftCalc[13] = wc * fftAddAvg(111,147); // 2220 - 2960 - fftCalc[14] = wc * fftAddAvg(147,194); // 2940 - 3900 - fftCalc[15] = wc * fftAddAvg(194,250); // 3880 - 5000 // avoid the last 5 bins, which are usually inaccurate -#else - //WLEDMM: different distributions - if (freqDist == 0) { - /* new mapping, optimized for 22050 Hz by softhack007 --- update: removed overlap */ - // bins frequency range - if (useInputFilter==1) { - // skip frequencies below 100hz - fftCalc[ 0] = wc * 0.8f * fftAddAvg(3,3); - fftCalc[ 1] = wc * 0.9f * fftAddAvg(4,4); - fftCalc[ 2] = wc * fftAddAvg(5,5); - fftCalc[ 3] = wc * fftAddAvg(6,6); - // don't use the last bins from 206 to 255. - fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping - } else { - fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass - fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass - fftCalc[ 2] = wc * fftAddAvg(3,4); // 2 129 - 216 bass - fftCalc[ 3] = wc * fftAddAvg(5,6); // 2 216 - 301 bass + midrange - // don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise) - fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping - } - fftCalc[ 4] = wc * fftAddAvg(7,9); // 3 301 - 430 midrange - fftCalc[ 5] = wc * fftAddAvg(10,12); // 3 430 - 560 midrange - fftCalc[ 6] = wc * fftAddAvg(13,18); // 5 560 - 818 midrange - fftCalc[ 7] = wc * fftAddAvg(19,25); // 7 818 - 1120 midrange -- 1Khz should always be the center ! - fftCalc[ 8] = wc * fftAddAvg(26,32); // 7 1120 - 1421 midrange - fftCalc[ 9] = wc * fftAddAvg(33,43); // 9 1421 - 1895 midrange - fftCalc[10] = wc * fftAddAvg(44,55); // 12 1895 - 2412 midrange + high mid - fftCalc[11] = wc * fftAddAvg(56,69); // 14 2412 - 3015 high mid - fftCalc[12] = wc * fftAddAvg(70,85); // 16 3015 - 3704 high mid - fftCalc[13] = wc * fftAddAvg(86,103); // 18 3704 - 4479 high mid - fftCalc[14] = wc * fftAddAvg(104,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping - } - else if (freqDist == 1) { //WLEDMM: Rightshift: note ewowi: frequencies in comments are not correct - if (useInputFilter==1) { - // skip frequencies below 100hz - fftCalc[ 0] = wc * 0.8f * fftAddAvg(1,1); - fftCalc[ 1] = wc * 0.9f * fftAddAvg(2,2); - fftCalc[ 2] = wc * fftAddAvg(3,3); - fftCalc[ 3] = wc * fftAddAvg(4,4); - // don't use the last bins from 206 to 255. - fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping - } else { - fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass - fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass - fftCalc[ 2] = wc * fftAddAvg(3,3); // 2 129 - 216 bass - fftCalc[ 3] = wc * fftAddAvg(4,4); // 2 216 - 301 bass + midrange - // don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise) - fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping - } - fftCalc[ 4] = wc * fftAddAvg(5,6); // 3 301 - 430 midrange - fftCalc[ 5] = wc * fftAddAvg(7,8); // 3 430 - 560 midrange - fftCalc[ 6] = wc * fftAddAvg(9,10); // 5 560 - 818 midrange - fftCalc[ 7] = wc * fftAddAvg(11,13); // 7 818 - 1120 midrange -- 1Khz should always be the center ! - fftCalc[ 8] = wc * fftAddAvg(14,18); // 7 1120 - 1421 midrange - fftCalc[ 9] = wc * fftAddAvg(19,25); // 9 1421 - 1895 midrange - fftCalc[10] = wc * fftAddAvg(26,36); // 12 1895 - 2412 midrange + high mid - fftCalc[11] = wc * fftAddAvg(37,45); // 14 2412 - 3015 high mid - fftCalc[12] = wc * fftAddAvg(46,66); // 16 3015 - 3704 high mid - fftCalc[13] = wc * fftAddAvg(67,97); // 18 3704 - 4479 high mid - fftCalc[14] = wc * fftAddAvg(98,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping - } -#endif - } else { // noise gate closed - just decay old values - isFirstRun = false; - 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; + //WLEDMM: different distributions + if (freqDist == 0) { + /* new mapping, optimized for 22050 Hz by softhack007 --- update: removed overlap */ + // bins frequency range + if (useInputFilter==1) { + // skip frequencies below 100hz + fftCalc[ 0] = wc * 0.8f * fftAddAvg(3,3); + fftCalc[ 1] = wc * 0.9f * fftAddAvg(4,4); + fftCalc[ 2] = wc * fftAddAvg(5,5); + fftCalc[ 3] = wc * fftAddAvg(6,6); + // don't use the last bins from 206 to 255. + fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping + } else { + fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass + fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass + fftCalc[ 2] = wc * fftAddAvg(3,4); // 2 129 - 216 bass + fftCalc[ 3] = wc * fftAddAvg(5,6); // 2 216 - 301 bass + midrange + // don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise) + fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping + } + fftCalc[ 4] = wc * fftAddAvg(7,9); // 3 301 - 430 midrange + fftCalc[ 5] = wc * fftAddAvg(10,12); // 3 430 - 560 midrange + fftCalc[ 6] = wc * fftAddAvg(13,18); // 5 560 - 818 midrange + fftCalc[ 7] = wc * fftAddAvg(19,25); // 7 818 - 1120 midrange -- 1Khz should always be the center ! + fftCalc[ 8] = wc * fftAddAvg(26,32); // 7 1120 - 1421 midrange + fftCalc[ 9] = wc * fftAddAvg(33,43); // 9 1421 - 1895 midrange + fftCalc[10] = wc * fftAddAvg(44,55); // 12 1895 - 2412 midrange + high mid + fftCalc[11] = wc * fftAddAvg(56,69); // 14 2412 - 3015 high mid + fftCalc[12] = wc * fftAddAvg(70,85); // 16 3015 - 3704 high mid + fftCalc[13] = wc * fftAddAvg(86,103); // 18 3704 - 4479 high mid + fftCalc[14] = wc * fftAddAvg(104,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping + } else if (freqDist == 1) { //WLEDMM: Rightshift: note ewowi: frequencies in comments are not correct + if (useInputFilter==1) { + // skip frequencies below 100hz + fftCalc[ 0] = wc * 0.8f * fftAddAvg(1,1); + fftCalc[ 1] = wc * 0.9f * fftAddAvg(2,2); + fftCalc[ 2] = wc * fftAddAvg(3,3); + fftCalc[ 3] = wc * fftAddAvg(4,4); + // don't use the last bins from 206 to 255. + fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping + } else { + fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass + fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass + fftCalc[ 2] = wc * fftAddAvg(3,3); // 2 129 - 216 bass + fftCalc[ 3] = wc * fftAddAvg(4,4); // 2 216 - 301 bass + midrange + // don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise) + fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping } + fftCalc[ 4] = wc * fftAddAvg(5,6); // 3 301 - 430 midrange + fftCalc[ 5] = wc * fftAddAvg(7,8); // 3 430 - 560 midrange + fftCalc[ 6] = wc * fftAddAvg(9,10); // 5 560 - 818 midrange + fftCalc[ 7] = wc * fftAddAvg(11,13); // 7 818 - 1120 midrange -- 1Khz should always be the center ! + fftCalc[ 8] = wc * fftAddAvg(14,18); // 7 1120 - 1421 midrange + fftCalc[ 9] = wc * fftAddAvg(19,25); // 9 1421 - 1895 midrange + fftCalc[10] = wc * fftAddAvg(26,36); // 12 1895 - 2412 midrange + high mid + fftCalc[11] = wc * fftAddAvg(37,45); // 14 2412 - 3015 high mid + fftCalc[12] = wc * fftAddAvg(46,66); // 16 3015 - 3704 high mid + fftCalc[13] = wc * fftAddAvg(67,97); // 18 3704 - 4479 high mid + fftCalc[14] = wc * fftAddAvg(98,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping } + } else { // noise gate closed - just decay old values + isFirstRun = false; + 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; + } } memcpy(lastFftCalc, fftCalc, sizeof(lastFftCalc)); // make a backup of last "good" channels @@ -1211,7 +1180,6 @@ class AudioReactive : public Usermod { 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 controller. double micLev = 0.0; // Used to convert returned value to have '0' as minimum. A leveller float expAdjF = 0.0f; // Used for exponential filter. @@ -1268,7 +1236,6 @@ class AudioReactive : public Usermod { //PLOT_PRINT("sampleAgc:"); PLOT_PRINT(sampleAgc); PLOT_PRINT("\t"); //PLOT_PRINT("sampleAvg:"); PLOT_PRINT(sampleAvg); PLOT_PRINT("\t"); //PLOT_PRINT("sampleReal:"); PLOT_PRINT(sampleReal); PLOT_PRINT("\t"); - //PLOT_PRINT("micIn:"); PLOT_PRINT(micIn); PLOT_PRINT("\t"); //PLOT_PRINT("sample:"); PLOT_PRINT(sample); PLOT_PRINT("\t"); //PLOT_PRINT("sampleMax:"); PLOT_PRINT(sampleMax); PLOT_PRINT("\t"); //PLOT_PRINT("multAgc:"); PLOT_PRINT(multAgc, 4); PLOT_PRINT("\t"); @@ -1418,6 +1385,15 @@ class AudioReactive : public Usermod { // update global vars ONCE - multAgc, sampleAGC, rawSampleAgc multAgc = multAgcTemp; + if (micQuality > 0) { + if (micQuality > 1) { + rawSampleAgc = 0.95f * tmpAgc + 0.05f * (float)rawSampleAgc; // raw path + sampleAgc += 0.95f * (tmpAgc - sampleAgc); // smooth path + } else { + rawSampleAgc = 0.70f * tmpAgc + 0.30f * (float)rawSampleAgc; // min filtering path + sampleAgc += 0.70f * (tmpAgc - sampleAgc); + } + } else { #if defined(WLEDMM_FASTPATH) rawSampleAgc = 0.65f * tmpAgc + 0.35f * (float)rawSampleAgc; #else @@ -1428,7 +1404,7 @@ class AudioReactive : public Usermod { sampleAgc = 0.5f * tmpAgc + 0.5f * sampleAgc; // fast path to zero else sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path - + } sampleAgc = fabsf(sampleAgc); // // make sure we have a positive value last_soundAgc = soundAgc; } // agcAvg() @@ -1446,27 +1422,6 @@ class AudioReactive : public Usermod { static unsigned long lastSoundTime = 0; // for delaying un-freeze static unsigned long startuptime = 0; // "fast freeze" mode: do not interfere during first 12 seconds (filter startup time) - #ifdef WLED_DISABLE_SOUND - micIn = inoise8(millis(), millis()); // Simulated analog read - micDataReal = micIn; - #else - #ifdef ARDUINO_ARCH_ESP32 - micIn = int(micDataReal); // micDataSm = ((micData * 3) + micData)/4; - #else - // this is the minimal code for reading analog mic input on 8266. - // warning!! Absolutely experimental code. Audio on 8266 is still not working. Expects a million follow-on problems. - static unsigned long lastAnalogTime = 0; - static float lastAnalogValue = 0.0f; - if (millis() - lastAnalogTime > 20) { - micDataReal = analogRead(A0); // read one sample with 10bit resolution. This is a dirty hack, supporting volumereactive effects only. - lastAnalogTime = millis(); - lastAnalogValue = micDataReal; - yield(); - } else micDataReal = lastAnalogValue; - micIn = int(micDataReal); - #endif - #endif - if (startuptime == 0) startuptime = millis(); // fast freeze mode - remember filter startup time if ((micLevelMethod < 1) || !isFrozen) { // following the input level, UNLESS mic Level was frozen micLev += (micDataReal-micLev) / 12288.0f; @@ -1477,7 +1432,6 @@ class AudioReactive : public Usermod { if (!haveSilence) isFrozen = true; // freeze mode: freeze micLevel so it cannot rise again } - micIn -= micLev; // Let's center it to 0 now // Using an exponential filter to smooth out the signal. We'll add controls for this in a future release. float micInNoDC = fabsf(micDataReal - micLev); @@ -1511,10 +1465,15 @@ class AudioReactive : public Usermod { if ((micLevelMethod == 2) && (millis() - startuptime < 12000)) isFrozen = false; // fast freeze: no freeze in first 12 seconds (filter startup phase) tmpSample = expAdjF; - micIn = abs(micIn); // And get the absolute value of each sample - sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // Adjust the gain. with inputLevel adjustment - sampleReal = tmpSample; + // Adjust the gain. with inputLevel adjustment. + if (micQuality > 0) { + sampleAdj = micInNoDC * sampleGain / 40.0f * inputLevel/128.0f + micInNoDC / 16.0f; // ... using unfiltered sample + sampleReal = micInNoDC; + } else { + sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // ... using pre-filtered sample + sampleReal = tmpSample; + } sampleAdj = fmax(fmin(sampleAdj, 255.0f), 0.0f); // Question: why are we limiting the value to 8 bits ??? sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!! @@ -1538,11 +1497,16 @@ class AudioReactive : public Usermod { } if (sampleMax < 0.5f) sampleMax = 0.0f; + if (micQuality > 0) { + if (micQuality > 1) sampleAvg += 0.95f * (sampleAdj - sampleAvg); + else sampleAvg += 0.70f * (sampleAdj - sampleAvg); + } else { #if defined(WLEDMM_FASTPATH) sampleAvg = ((sampleAvg * 11.0f) + sampleAdj) / 12.0f; // make reactions a bit more "crisp" in fastpath mode #else sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples. #endif + } sampleAvg = fabsf(sampleAvg); // make sure we have a positive value } // getSample() @@ -2784,6 +2748,7 @@ class AudioReactive : public Usermod { //WLEDMM: experimental settings JsonObject poweruser = top.createNestedObject("experiments"); poweruser[F("micLev")] = micLevelMethod; + poweruser[F("Mic_Quality")] = micQuality; poweruser[F("freqDist")] = freqDist; //poweruser[F("freqRMS")] = averageByRMS; poweruser[F("FFT_Window")] = fftWindow; @@ -2859,6 +2824,7 @@ class AudioReactive : public Usermod { //WLEDMM: experimental settings configComplete &= getJsonValue(top["experiments"][F("micLev")], micLevelMethod); + configComplete &= getJsonValue(top["experiments"][F("Mic_Quality")], micQuality); configComplete &= getJsonValue(top["experiments"][F("freqDist")], freqDist); //configComplete &= getJsonValue(top["experiments"][F("freqRMS")], averageByRMS); configComplete &= getJsonValue(top["experiments"][F("FFT_Window")], fftWindow); @@ -2959,23 +2925,29 @@ class AudioReactive : public Usermod { oappend(SET_F("addOption(dd,'Lazy',3);")); //WLEDMM: experimental settings - oappend(SET_F("dd=addDropdown(ux,'experiments:micLev');")); + oappend(SET_F("xx='experiments';")); // shortcut + oappend(SET_F("dd=addDropdown(ux,xx+':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(ux+':experiments:micLev',1,'☾');")); + oappend(SET_F("addInfo(ux+':'+xx+':micLev',1,'☾');")); - oappend(SET_F("dd=addDropdown(ux,'experiments:freqDist');")); + oappend(SET_F("dd=addDropdown(ux,xx+':Mic_Quality');")); + oappend(SET_F("addOption(dd,'average (standard)',0);")); + oappend(SET_F("addOption(dd,'low noise',1);")); + oappend(SET_F("addOption(dd,'perfect',2);")); + + oappend(SET_F("dd=addDropdown(ux,xx+':freqDist');")); oappend(SET_F("addOption(dd,'Normal (⎌)',0);")); oappend(SET_F("addOption(dd,'RightShift',1);")); - oappend(SET_F("addInfo(ux+':experiments:freqDist',1,'☾');")); + oappend(SET_F("addInfo(ux+':'+xx+':freqDist',1,'☾');")); - //oappend(SET_F("dd=addDropdown(ux,'experiments:freqRMS');")); + //oappend(SET_F("dd=addDropdown(ux,xx+':freqRMS');")); //oappend(SET_F("addOption(dd,'Off (⎌)',0);")); //oappend(SET_F("addOption(dd,'On',1);")); //oappend(SET_F("addInfo(ux+':experiments:freqRMS',1,'☾');")); - oappend(SET_F("dd=addDropdown(ux,'experiments:FFT_Window');")); + oappend(SET_F("dd=addDropdown(ux,xx+':FFT_Window');")); oappend(SET_F("addOption(dd,'Blackman-Harris (MM standard)',0);")); oappend(SET_F("addOption(dd,'Hann (balanced)',1);")); oappend(SET_F("addOption(dd,'Nuttall (more accurate)',2);")); @@ -2984,10 +2956,10 @@ class AudioReactive : public Usermod { oappend(SET_F("addOption(dd,'Flat-Top (AC WLED, inaccurate)',4);")); #ifdef FFT_USE_SLIDING_WINDOW - oappend(SET_F("dd=addDropdown(ux,'experiments:I2S_FastPath');")); + oappend(SET_F("dd=addDropdown(ux,xx+':I2S_FastPath');")); oappend(SET_F("addOption(dd,'Off',0);")); oappend(SET_F("addOption(dd,'On (⎌)',1);")); - oappend(SET_F("addInfo(ux+':experiments:I2S_FastPath',1,'☾');")); + oappend(SET_F("addInfo(ux+':'+xx+':I2S_FastPath',1,'☾');")); #endif oappend(SET_F("dd=addDropdown(ux,'dynamics:limiter');")); From 528ec1fa4cfb46025519183c387eafdc3ce3c262 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Fri, 5 Jul 2024 18:25:48 +0200 Subject: [PATCH 27/29] version of today, -b32.41 featuring FastPath v2 --- package-lock.json | 4 ++-- package.json | 2 +- wled00/wled.h | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/package-lock.json b/package-lock.json index 80dcf5ec..dce951b8 100644 --- a/package-lock.json +++ b/package-lock.json @@ -1,12 +1,12 @@ { "name": "wled", - "version": "0.14.1-b32.40", + "version": "0.14.1-b32.41", "lockfileVersion": 2, "requires": true, "packages": { "": { "name": "wled", - "version": "0.14.1-b32.40", + "version": "0.14.1-b32.41", "license": "GPL-3.0-or-later", "dependencies": { "clean-css": "^4.2.3", diff --git a/package.json b/package.json index 0536796c..2faab264 100644 --- a/package.json +++ b/package.json @@ -1,6 +1,6 @@ { "name": "wled", - "version": "0.14.1-b32.40", + "version": "0.14.1-b32.41", "description": "Tools for WLED project", "main": "tools/cdata.js", "directories": { diff --git a/wled00/wled.h b/wled00/wled.h index 8ff90dd5..f06f3a36 100644 --- a/wled00/wled.h +++ b/wled00/wled.h @@ -8,7 +8,7 @@ */ // version code in format yymmddb (b = daily build) -#define VERSION 2406240 +#define VERSION 2407050 // WLEDMM - you can check for this define in usermods, to only enabled WLEDMM specific code in the "right" fork. Its not defined in AC WLED. #define _MoonModules_WLED_ From 5123128e9c7bcc69a2397a0bd36ebb566548e481 Mon Sep 17 00:00:00 2001 From: Will Tatam Date: Tue, 9 Jul 2024 18:39:58 +0100 Subject: [PATCH 28/29] Limit hub75 chain length by height 64 not width to allow for 2 x 64x32 --- wled00/bus_manager.cpp | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/wled00/bus_manager.cpp b/wled00/bus_manager.cpp index fff5ecb3..cb00ad29 100644 --- a/wled00/bus_manager.cpp +++ b/wled00/bus_manager.cpp @@ -518,9 +518,7 @@ BusHub75Matrix::BusHub75Matrix(BusConfig &bc) : Bus(bc.type, bc.start, bc.autoWh break; } - mxconfig.chain_length = max((u_int8_t) 1, min(bc.pins[0], (u_int8_t) 4)); // prevent bad data preventing boot due to low memory - - if(mxconfig.mx_width >= 64 && (bc.pins[0] > 1)) { + if(mxconfig.mx_height >= 64 && (bc.pins[0] > 1)) { USER_PRINT("WARNING, only single panel can be used of 64 pixel boards due to memory") mxconfig.chain_length = 1; } @@ -609,6 +607,7 @@ BusHub75Matrix::BusHub75Matrix(BusConfig &bc) : Bus(bc.type, bc.start, bc.autoWh #endif + mxconfig.chain_length = max((u_int8_t) 1, min(bc.pins[0], (u_int8_t) 4)); // prevent bad data preventing boot due to low memory USER_PRINTF("MatrixPanel_I2S_DMA config - %ux%u length: %u\n", mxconfig.mx_width, mxconfig.mx_height, mxconfig.chain_length); From fb30f9c64142dad49cdda3baa73f959e52969929 Mon Sep 17 00:00:00 2001 From: Will Tatam Date: Tue, 9 Jul 2024 21:35:53 +0100 Subject: [PATCH 29/29] Add Hub75Matrix 64x32 (Outdoor 8S) --- wled00/bus_manager.cpp | 8 +++++++- wled00/data/settings_leds.htm | 3 ++- 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/wled00/bus_manager.cpp b/wled00/bus_manager.cpp index cb00ad29..2bd0a338 100644 --- a/wled00/bus_manager.cpp +++ b/wled00/bus_manager.cpp @@ -650,12 +650,18 @@ BusHub75Matrix::BusHub75Matrix(BusConfig &bc) : Bus(bc.type, bc.start, bc.autoWh switch(bc.type) { case 105: - USER_PRINTLN("MatrixPanel_I2S_DMA FOUR_SCAN_32PX_HIGH"); + USER_PRINTLN("MatrixPanel_I2S_DMA FOUR_SCAN_32PX_HIGH - 32x32"); fourScanPanel = new VirtualMatrixPanel((*display), 1, 1, 32, 32); fourScanPanel->setPhysicalPanelScanRate(FOUR_SCAN_32PX_HIGH); fourScanPanel->setRotation(0); break; case 106: + USER_PRINTLN("MatrixPanel_I2S_DMA FOUR_SCAN_32PX_HIGH - 64x32"); + fourScanPanel = new VirtualMatrixPanel((*display), 1, 1, 64, 32); + fourScanPanel->setPhysicalPanelScanRate(FOUR_SCAN_32PX_HIGH); + fourScanPanel->setRotation(0); + break; + case 107: USER_PRINTLN("MatrixPanel_I2S_DMA FOUR_SCAN_64PX_HIGH"); fourScanPanel = new VirtualMatrixPanel((*display), 1, 1, 64, 64); fourScanPanel->setPhysicalPanelScanRate(FOUR_SCAN_64PX_HIGH); diff --git a/wled00/data/settings_leds.htm b/wled00/data/settings_leds.htm index e1979fdc..5e31ee4c 100644 --- a/wled00/data/settings_leds.htm +++ b/wled00/data/settings_leds.htm @@ -384,7 +384,8 @@ ${i+1}: - + +
Color Order: