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prepare for merge with upstream

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- prepare for merging upstream
--> audiorective files should stay the same after merging!
- makuna just release NeoPixelBus 2.7.1 -> use for "V4" build
This commit is contained in:
Frank
2022-11-29 13:49:17 +01:00
parent 8f4c6d625d
commit 1f0b53ce7a
4 changed files with 108 additions and 58 deletions
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4
platformio.ini
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@@ -268,7 +268,7 @@ build_flagsV4 = -g
;;; V4.4.x libraries (without LOROL_LITTLEFS; with newer NeoPixelBus) ;;; V4.4.x libraries (without LOROL_LITTLEFS; with newer NeoPixelBus)
lib_depsV4 = lib_depsV4 =
${env.lib_deps} ${env.lib_deps}
https://github.com/Makuna/NeoPixelBus.git#master @ 2.7.0 ;; NPB 2.6.9 tends to crash whith IDF V4.4.3 -> use latest NeoPixelBus dev version instead https://github.com/Makuna/NeoPixelBus.git#master @ 2.7.1 ;; NPB 2.6.9 tends to crash whith IDF V4.4.3 -> use latest NeoPixelBus 2.7.1 instead
https://github.com/pbolduc/AsyncTCP.git @ 1.2.0 https://github.com/pbolduc/AsyncTCP.git @ 1.2.0
[esp32s2] [esp32s2]
@@ -310,7 +310,7 @@ build_flags = -g
lib_deps = lib_deps =
${env.lib_deps} ${env.lib_deps}
;; currently we need the latest NeoPixelBus dev version, because it contains important bugfixes for -S3 ;; currently we need the latest NeoPixelBus dev version, because it contains important bugfixes for -S3
https://github.com/Makuna/NeoPixelBus.git#master @ 2.7.0 https://github.com/Makuna/NeoPixelBus.git#master @ 2.7.1
https://github.com/pbolduc/AsyncTCP.git @ 1.2.0 https://github.com/pbolduc/AsyncTCP.git @ 1.2.0

88
usermods/audioreactive/audio_reactive.h
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@@ -54,6 +54,16 @@
#endif #endif
#endif #endif
#if defined(MIC_LOGGER) || defined(FFT_SAMPLING_LOG)
#define PLOT_PRINT(x) DEBUGOUT.print(x)
#define PLOT_PRINTLN(x) DEBUGOUT.println(x)
#define PLOT_PRINTF(x...) DEBUGOUT.printf(x)
#else
#define PLOT_PRINT(x)
#define PLOT_PRINTLN(x)
#define PLOT_PRINTF(x...)
#endif
// use audio source class (ESP32 specific) // use audio source class (ESP32 specific)
#include "audio_source.h" #include "audio_source.h"
constexpr i2s_port_t I2S_PORT = I2S_NUM_0; // I2S port to use (do not change !) constexpr i2s_port_t I2S_PORT = I2S_NUM_0; // I2S port to use (do not change !)
@@ -137,6 +147,8 @@ static void autoResetPeak(void); // peak auto-reset function
//////////////////// ////////////////////
// some prototypes, to ensure consistent interfaces // some prototypes, to ensure consistent interfaces
static float mapf(float x, float in_min, float in_max, float out_min, float out_max); // map function for float
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 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 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); // post-processing and post-amp of GEQ channels
@@ -228,7 +240,7 @@ constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT resul
// the following are observed values, supported by a bit of "educated guessing" // 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 0.65f // 20kHz - downscaling factor for FFT results - "Flat-Top" window @20Khz, old freq channels
#define FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels #define FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels
#define LOG_256 5.54517744 // log2(256) #define LOG_256 5.54517744f // log(256)
// These are the input and output vectors. Input vectors receive computed results from FFT. // These are the input and output vectors. Input vectors receive computed results from FFT.
static float vReal[samplesFFT] = {0.0f}; // FFT sample inputs / freq output - these are our raw result bins static float vReal[samplesFFT] = {0.0f}; // FFT sample inputs / freq output - these are our raw result bins
@@ -247,6 +259,7 @@ static float windowWeighingFactors[samplesFFT] = {0.0f};
// lib_deps += https://github.com/blazoncek/arduinoFFT.git // lib_deps += https://github.com/blazoncek/arduinoFFT.git
#endif #endif
#include <arduinoFFT.h> #include <arduinoFFT.h>
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT #ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, windowWeighingFactors); static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, windowWeighingFactors);
#else #else
@@ -488,10 +501,10 @@ static void runMicFilter(uint16_t numSamples, float *sampleBuffer) // p
constexpr float alpha = 0.0225f; // 80hz constexpr float alpha = 0.0225f; // 80hz
//constexpr float alpha = 0.01693f;// 60hz //constexpr float alpha = 0.01693f;// 60hz
// high frequency cutoff parameter // high frequency cutoff parameter
//constexpr float beta1 = 0.75; // 11Khz //constexpr float beta1 = 0.75f; // 11Khz
//constexpr float beta1 = 0.82; // 15Khz //constexpr float beta1 = 0.82f; // 15Khz
//constexpr float beta1 = 0.8285; // 18Khz //constexpr float beta1 = 0.8285f; // 18Khz
constexpr float beta1 = 0.85; // 20Khz constexpr float beta1 = 0.85f; // 20Khz
constexpr float beta2 = (1.0f - beta1) / 2.0; constexpr float beta2 = (1.0f - beta1) / 2.0;
static float last_vals[2] = { 0.0f }; // FIR high freq cutoff filter static float last_vals[2] = { 0.0f }; // FIR high freq cutoff filter
@@ -620,7 +633,6 @@ static void detectSamplePeak(void) {
timeOfPeak = millis(); timeOfPeak = millis();
udpSamplePeak = true; udpSamplePeak = true;
} }
} }
static void autoResetPeak(void) { static void autoResetPeak(void) {
@@ -724,7 +736,7 @@ class AudioReactive : public Usermod {
// variables used by getSample() and agcAvg() // 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 int16_t micIn = 0; // Current sample starts with negative values and large values, which is why it's 16 bit signed
double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controler. double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controler.
double micLev = 0.0f; // Used to convert returned value to have '0' as minimum. A leveller double micLev = 0.0; // Used to convert returned value to have '0' as minimum. A leveller
float expAdjF = 0.0f; // Used for exponential filter. float expAdjF = 0.0f; // Used for exponential filter.
float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC. float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC.
int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel) int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel)
@@ -760,29 +772,29 @@ class AudioReactive : public Usermod {
if (disableSoundProcessing && (!udpSyncConnected || ((audioSyncEnabled & 0x02) == 0))) return; // no audio availeable if (disableSoundProcessing && (!udpSyncConnected || ((audioSyncEnabled & 0x02) == 0))) return; // no audio availeable
#ifdef MIC_LOGGER #ifdef MIC_LOGGER
// Debugging functions for audio input and sound processing. Comment out the values you want to see // Debugging functions for audio input and sound processing. Comment out the values you want to see
Serial.print("micReal:"); Serial.print(micDataReal); Serial.print("\t"); PLOT_PRINT("micReal:"); PLOT_PRINT(micDataReal); PLOT_PRINT("\t");
Serial.print("volumeSmth:"); Serial.print(volumeSmth); Serial.print("\t"); PLOT_PRINT("volumeSmth:"); PLOT_PRINT(volumeSmth); PLOT_PRINT("\t");
//Serial.print("volumeRaw:"); Serial.print(volumeRaw); Serial.print("\t"); //PLOT_PRINT("volumeRaw:"); PLOT_PRINT(volumeRaw); PLOT_PRINT("\t");
Serial.print("DC_Level:"); Serial.print(micLev); Serial.print("\t"); PLOT_PRINT("DC_Level:"); PLOT_PRINT(micLev); PLOT_PRINT("\t");
//Serial.print("sampleAgc:"); Serial.print(sampleAgc); Serial.print("\t"); //PLOT_PRINT("sampleAgc:"); PLOT_PRINT(sampleAgc); PLOT_PRINT("\t");
//Serial.print("sampleAvg:"); Serial.print(sampleAvg); Serial.print("\t"); //PLOT_PRINT("sampleAvg:"); PLOT_PRINT(sampleAvg); PLOT_PRINT("\t");
//Serial.print("sampleReal:"); Serial.print(sampleReal); Serial.print("\t"); //PLOT_PRINT("sampleReal:"); PLOT_PRINT(sampleReal); PLOT_PRINT("\t");
//Serial.print("micIn:"); Serial.print(micIn); Serial.print("\t"); //PLOT_PRINT("micIn:"); PLOT_PRINT(micIn); PLOT_PRINT("\t");
//Serial.print("sample:"); Serial.print(sample); Serial.print("\t"); //PLOT_PRINT("sample:"); PLOT_PRINT(sample); PLOT_PRINT("\t");
//Serial.print("sampleMax:"); Serial.print(sampleMax); Serial.print("\t"); //PLOT_PRINT("sampleMax:"); PLOT_PRINT(sampleMax); PLOT_PRINT("\t");
//Serial.print("samplePeak:"); Serial.print((samplePeak!=0) ? 128:0); Serial.print("\t"); //PLOT_PRINT("samplePeak:"); PLOT_PRINT((samplePeak!=0) ? 128:0); PLOT_PRINT("\t");
//Serial.print("multAgc:"); Serial.print(multAgc, 4); Serial.print("\t"); //PLOT_PRINT("multAgc:"); PLOT_PRINT(multAgc, 4); PLOT_PRINT("\t");
Serial.println(); PLOT_PRINTLN();
#endif #endif
#ifdef FFT_SAMPLING_LOG #ifdef FFT_SAMPLING_LOG
#if 0 #if 0
for(int i=0; i<NUM_GEQ_CHANNELS; i++) { for(int i=0; i<NUM_GEQ_CHANNELS; i++) {
Serial.print(fftResult[i]); PLOT_PRINT(fftResult[i]);
Serial.print("\t"); PLOT_PRINT("\t");
} }
Serial.println(); PLOT_PRINTLN();
#endif #endif
// OPTIONS are in the following format: Description \n Option // OPTIONS are in the following format: Description \n Option
// //
@@ -808,20 +820,21 @@ class AudioReactive : public Usermod {
if(fftResult[i] < minVal) minVal = fftResult[i]; if(fftResult[i] < minVal) minVal = fftResult[i];
} }
for(int i = 0; i < NUM_GEQ_CHANNELS; i++) { for(int i = 0; i < NUM_GEQ_CHANNELS; i++) {
Serial.print(i); Serial.print(":"); PLOT_PRINT(i); PLOT_PRINT(":");
Serial.printf("%04ld ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1)); PLOT_PRINTF("%04ld ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1));
} }
if(printMaxVal) { if(printMaxVal) {
Serial.printf("maxVal:%04d ", maxVal + (mapValuesToPlotterSpace ? 16*256 : 0)); PLOT_PRINTF("maxVal:%04d ", maxVal + (mapValuesToPlotterSpace ? 16*256 : 0));
} }
if(printMinVal) { if(printMinVal) {
Serial.printf("%04d:minVal ", minVal); // printed with value first, then label, so negative values can be seen in Serial Monitor but don't throw off y axis in Serial Plotter PLOT_PRINTF("%04d:minVal ", minVal); // printed with value first, then label, so negative values can be seen in Serial Monitor but don't throw off y axis in Serial Plotter
} }
if(mapValuesToPlotterSpace) if(mapValuesToPlotterSpace)
Serial.printf("max:%04d ", (printMaxVal ? 17 : 16)*256); // print line above the maximum value we expect to see on the plotter to avoid autoscaling y axis PLOT_PRINTF("max:%04d ", (printMaxVal ? 17 : 16)*256); // print line above the maximum value we expect to see on the plotter to avoid autoscaling y axis
else else {
Serial.printf("max:%04d ", 256); PLOT_PRINTF("max:%04d ", 256);
Serial.println(); }
PLOT_PRINTLN();
#endif // FFT_SAMPLING_LOG #endif // FFT_SAMPLING_LOG
} // logAudio() } // logAudio()
@@ -955,12 +968,7 @@ class AudioReactive : public Usermod {
#endif #endif
#endif #endif
//micLev = ((micLev * 8191.0f) + micDataReal) / 8192.0f; // takes a few seconds to "catch up" with the Mic Input
//if (useBandPassFilter)
// micLev += (micDataReal-micLev) / 8192.0f; // we expect some more fluctuations with mics that need pre-filtering
//else
micLev += (micDataReal-micLev) / 12288.0f; micLev += (micDataReal-micLev) / 12288.0f;
if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal
micIn -= micLev; // Let's center it to 0 now micIn -= micLev; // Let's center it to 0 now
@@ -1767,7 +1775,7 @@ class AudioReactive : public Usermod {
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
JsonObject amic = top.createNestedObject(FPSTR(_analogmic)); JsonObject amic = top.createNestedObject(FPSTR(_analogmic));
amic["pin"] = audioPin; amic["pin"] = audioPin;
#endif #endif
JsonObject dmic = top.createNestedObject(FPSTR(_digitalmic)); JsonObject dmic = top.createNestedObject(FPSTR(_digitalmic));
dmic[F("type")] = dmType; dmic[F("type")] = dmType;
@@ -1918,8 +1926,8 @@ class AudioReactive : public Usermod {
#else #else
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',3,'', 'I2S Master CLK');")); oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',3,'', 'I2S Master CLK');"));
#endif #endif
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',4,'I2C SDA',' ');")); oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',4,'', 'I2C SDA');"));
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',5,'I2C SCL',' ');")); oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',5,'', 'I2C SCL');"));
} }

13
usermods/audioreactive/audio_source.h
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@@ -218,10 +218,15 @@ class I2SSource : public AudioSource {
#endif #endif
#endif #endif
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
// example from espressif: https://github.com/espressif/esp-idf/blob/release/v4.4/examples/peripherals/i2s/i2s_audio_recorder_sdcard/main/i2s_recorder_main.c
// This is an I2S PDM microphone, these microphones only use a clock and // This is an I2S PDM microphone, these microphones only use a clock and
// data line, to make it simpler to debug, use the WS pin as CLK and SD // data line, to make it simpler to debug, use the WS pin as CLK and SD pin as DATA
// pin as DATA // example from espressif: https://github.com/espressif/esp-idf/blob/release/v4.4/examples/peripherals/i2s/i2s_audio_recorder_sdcard/main/i2s_recorder_main.c
// note to self: PDM has known bugs on S3, and does not work on C3
// * S3: PDM sample rate only at 50% of expected rate: https://github.com/espressif/esp-idf/issues/9893
// * S3: I2S PDM has very low amplitude: https://github.com/espressif/esp-idf/issues/8660
// * C3: does not support PDM to PCM input. SoC would allow PDM RX, but there is no hardware to directly convert to PCM so it will not work. https://github.com/espressif/esp-idf/issues/8796
_config.mode = i2s_mode_t(I2S_MODE_MASTER | I2S_MODE_RX | I2S_MODE_PDM); // Change mode to pdm if clock pin not provided. PDM is not supported on ESP32-S2. PDM RX not supported on ESP32-C3 _config.mode = i2s_mode_t(I2S_MODE_MASTER | I2S_MODE_RX | I2S_MODE_PDM); // Change mode to pdm if clock pin not provided. PDM is not supported on ESP32-S2. PDM RX not supported on ESP32-C3
_config.channel_format =I2S_PDM_MIC_CHANNEL; // seems that PDM mono mode always uses left channel. _config.channel_format =I2S_PDM_MIC_CHANNEL; // seems that PDM mono mode always uses left channel.
_config.use_apll = true; // experimental - use aPLL clock source to improve sampling quality _config.use_apll = true; // experimental - use aPLL clock source to improve sampling quality
@@ -424,7 +429,7 @@ class ES7243 : public I2SSource {
Wire.write((uint8_t)val); Wire.write((uint8_t)val);
uint8_t i2cErr = Wire.endTransmission(); // i2cErr == 0 means OK uint8_t i2cErr = Wire.endTransmission(); // i2cErr == 0 means OK
if (i2cErr != 0) { if (i2cErr != 0) {
DEBUGSR_PRINTF("AR: ES7243 I2C write failed with error=%d (addr=0x%X, reg 0x%X, val 0x%X).\n", ES7243_ADDR, i2cErr, reg, val); DEBUGSR_PRINTF("AR: ES7243 I2C write failed with error=%d (addr=0x%X, reg 0x%X, val 0x%X).\n", i2cErr, ES7243_ADDR, reg, val);
} }
} }

61
usermods/audioreactive/readme.md
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@@ -1,36 +1,73 @@
# Audioreactive usermod # Audioreactive usermod
This usermod allows controlling LEDs using audio input. Audio input can be either microphone or analog-in (AUX) using appropriate adapter. This usermod allows controlling LEDs using audio input. Audio input can be either microphone or analog-in (AUX) using appropriate adapter.
Supported microphones range from analog (MAX4466, MAX9814, ...) to digital (INMP441, ICS-43434, ...). Supported microphones range from cheap analog (MAX4466, MAX9814, ...) to high quality digital (INMP441, ICS-43434, ...) and dgital Line-In.
The usermod does audio processing and provides data structure that specially written effect can use. The usermod does audio processing and provides data structure that specially written effect can use.
The usermod **does not** provide effects or draws anything to LED strip/matrix. The usermod **does not** provide effects or draws anything to LED strip/matrix.
## Additional Documentation
This usermod is an evolution of [SR-WLED](https://github.com/atuline/WLED), and a lot of documentation and information can be found in the [SR-WLED wiki](https://github.com/atuline/WLED/wiki):
* [getting started with audio](https://github.com/atuline/WLED/wiki/First-Time-Setup#sound)
* [Sound settings](https://github.com/atuline/WLED/wiki/Sound-Settings) - similar to options on the usemod settings page in WLED.
* [Digital Audio](https://github.com/atuline/WLED/wiki/Digital-Microphone-Hookup)
* [Analog Audio](https://github.com/atuline/WLED/wiki/Analog-Audio-Input-Options)
* [UDP Sound sync](https://github.com/atuline/WLED/wiki/UDP-Sound-Sync)
## Supported MCUs
This audioreactive usermod works best on "classic ESP32" (dual core), and on ESP32-S3 which also has dual core and hardware floating point support.
It will compile succesfully for ESP32-S2 and ESP32-C3, however might not work well, as other WLED functions will become slow. Audio processing requires a lot of computing power, which can be problematic on smaller MCUs like -S2 and -C3.
Analog audio is only possible on "classic" ESP32, but not on other MCUs like ESP32-S3.
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 ## Installation
Add `-D USERMOD_AUDIOREACTIVE` to your PlatformIO environment as well as `arduinoFFT` to your `lib_deps`. ### 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. 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.
Customised _arduinoFFT_ library for use with this usermod can be found at https://github.com/blazoncek/arduinoFFT.git 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 slighly faster than our customised library, however also needs additional 2kB RAM.
* `build_flags` = `-D USERMOD_AUDIOREACTIVE` `-D UM_AUDIOREACTIVE_USE_NEW_FFT`
* `lib_deps`= `https://github.com/kosme/arduinoFFT#develop @ 1.9.2`
## Configuration ## Configuration
All parameters are runtime configurable though some may require hard boot after change (I2S microphone or selected GPIOs). All parameters are runtime configurable though some may require hard boot after change (I2S microphone or selected GPIOs).
If you want to define default GPIOs during compile time use the following (default values in parentheses): If you want to define default GPIOs during compile time use the following addtional build_flags (default values in parentheses):
- `SR_DMTYPE=x` : defines digital microphone type: 0=analog, 1=generic I2S, 2=ES7243 I2S, 3=SPH0645 I2S, 4=generic I2S with master clock, 5=PDM I2S - `-D SR_DMTYPE=x` : defines digital microphone type: 0=analog, 1=generic I2S (default), 2=ES7243 I2S, 3=SPH0645 I2S, 4=generic I2S with master clock, 5=PDM I2S
- `AUDIOPIN=x` : GPIO for analog microphone/AUX-in (36) - `-D AUDIOPIN=x` : GPIO for analog microphone/AUX-in (36)
- `I2S_SDPIN=x` : GPIO for SD pin on digital mcrophone (32) - `-D I2S_SDPIN=x` : GPIO for SD pin on digital microphone (32)
- `I2S_WSPIN=x` : GPIO for WS pin on digital mcrophone (15) - `-D I2S_WSPIN=x` : GPIO for WS pin on digital microphone (15)
- `I2S_CKPIN=x` : GPIO for SCK pin on digital mcrophone (14) - `-D I2S_CKPIN=x` : GPIO for SCK pin on digital microphone (14)
- `ES7243_SDAPIN` : GPIO for I2C SDA pin on ES7243 microphone (-1) - `-D MCLK_PIN=x` : GPIO for master clock pin on digital Line-In boards (-1)
- `ES7243_SCLPIN` : GPIO for I2C SCL pin on ES7243 microphone (-1) - `-D ES7243_SDAPIN` : GPIO for I2C SDA pin on ES7243 microphone (-1)
- `MCLK_PIN=x` : GPIO for master clock pin on digital mcrophone (-1) - `-D ES7243_SCLPIN` : GPIO for I2C SCL pin on ES7243 microphone (-1)
**NOTE** Due to the fact that usermod uses I2S peripherial for analog audio sampling, use of analog *buttons* (i.e. potentiometers) is disabled while running this usermod with analog microphone. **NOTE** Due to the fact that usermod uses I2S peripherial for analog audio sampling, use of analog *buttons* (i.e. potentiometers) is disabled while running this usermod with analog microphone.
### Advanced Compile-Time Options
You can use the following additional flags in your `build_flags`
* `-D SR_SQUELCH=x` : Default "squelch" setting (10)
* `-D SR_GAIN=x` : Default "gain" setting (60)
* `-D I2S_USE_RIGHT_CHANNEL`: Use RIGHT instead of LEFT channel (not recommended unless you strictly need this).
* `-D I2S_USE_16BIT_SAMPLES`: Use 16bit instead of 32bit for internal sample buffers. Reduces sampling quality, but frees some RAM ressources (not recommended unless you absolutely need this).
* `-D I2S_GRAB_ADC1_COMPLETELY`: Experimental: continously sample analog ADC microphone. Only effective on ESP32. WARNING this _will_ cause conflicts(lock-up) with any analogRead() call.
* `-D MIC_LOGGER` : (debugging) Logs samples from the microphone to serial USB. Use with serial plotter (Arduino IDE)
* `-D SR_DEBUG` : (debugging) Additional error diagnostics and debug info on serial USB.
## Release notes ## Release notes
2022-06 Ported from [soundreactive](https://github.com/atuline/WLED) by @blazoncek (AKA Blaz Kristan) * 2022-06 Ported from [soundreactive WLED](https://github.com/atuline/WLED) - by @blazoncek (AKA Blaz Kristan) and the [SR-WLED team](https://github.com/atuline/WLED/wiki#sound-reactive-wled-fork-team).
* 2022-11 Updated to align with "[MoonModules/WLED](https://amg.wled.me)" audioreactive usermod - by @softhack007 (AKA Frank M&ouml;hle).