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AR: trying to improve coexistence with HUB75

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Hub75 is very memory hungry. So we try to make a bit more RAM available
* use 16bit samples when compiling with HUB75 support --> 3KB saved
* avoid using aPLL (HUB75 needs it)
* move audio buffers from BSS (always allocated) to heap (only allocated when FFTtask runs) --> 10Kb saved
* suspend live preview for 6 seconds when out-of-memory
This commit is contained in:
Frank
2024-10-13 23:06:50 +02:00
parent d21b9b0687
commit 0dd036cdce
3 changed files with 74 additions and 19 deletions
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72
usermods/audioreactive/audio_reactive.h
View File

@@ -284,7 +284,7 @@ static volatile float micReal_max2 = 0.0f; // MicIn data max afte
// 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) __attribute__((noreturn)); // 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 postProcessFFTResults(bool noiseGateOpen, int numberOfChannels, bool i2sFastpath); // post-processing and post-amp of GEQ channels
@@ -393,11 +393,11 @@ constexpr uint16_t samplesFFT_2 = 256; // meaningful part of FFT result
#define LOG_256 5.54517744f // log(256)
// 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 vImag[samplesFFT] = {0.0f}; // imaginary parts
static float* vReal = nullptr; // FFT sample inputs / freq output - these are our raw result bins
static float* vImag = nullptr; // imaginary parts
#ifdef FFT_MAJORPEAK_HUMAN_EAR
static float pinkFactors[samplesFFT] = {0.0f}; // "pink noise" correction factors
static float* pinkFactors = nullptr; // "pink noise" correction factors
constexpr float pinkcenter = 23.66; // sqrt(560) - center freq for scaling is 560 hz.
constexpr float binWidth = SAMPLE_RATE / (float)samplesFFT; // frequency range of each FFT result bin
#endif
@@ -414,15 +414,6 @@ constexpr float binWidth = SAMPLE_RATE / (float)samplesFFT; // frequency range o
#define sqrt_internal sqrtf // see https://github.com/kosme/arduinoFFT/pull/83
#include <arduinoFFT.h>
#if defined(FFT_LIB_REV) && FFT_LIB_REV > 0x19
// arduinoFFT 2.x has a slightly different API
static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, true);
#else
// recommended version optimized by @softhack007 (API version 1.9)
static float windowWeighingFactors[samplesFFT] = {0.0f}; // cache for FFT windowing factors
static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, windowWeighingFactors);
#endif
// Helper functions
// float version of map()
@@ -460,6 +451,30 @@ constexpr bool skipSecondFFT = true;
constexpr bool skipSecondFFT = false;
#endif
// allocate FFT sample buffers from heap
static bool alocateFFTBuffers(void) {
#ifdef SR_DEBUG
USER_PRINT(F("\nFree heap ")); USER_PRINTLN(ESP.getFreeHeap());
#endif
if (vReal) free(vReal); // should not happen
if (vImag) free(vImag); // should not happen
if ((vReal = (float*) calloc(sizeof(float), samplesFFT)) == nullptr) return false; // calloc or die
if ((vImag = (float*) calloc(sizeof(float), samplesFFT)) == nullptr) return false;
#ifdef FFT_MAJORPEAK_HUMAN_EAR
if (pinkFactors) free(pinkFactors);
if ((pinkFactors = (float*) calloc(sizeof(float), samplesFFT)) == nullptr) return false;
#endif
#ifdef SR_DEBUG
USER_PRINTLN("\nalocateFFTBuffers() completed successfully.");
USER_PRINT(F("Free heap: ")); USER_PRINTLN(ESP.getFreeHeap());
USER_PRINT("FFTtask free stack: "); USER_PRINTLN(uxTaskGetStackHighWaterMark(NULL));
USER_FLUSH();
#endif
return(true); // success
}
// High-Pass "DC blocker" filter
// see https://www.dsprelated.com/freebooks/filters/DC_Blocker.html
static void runDCBlocker(uint_fast16_t numSamples, float *sampleBuffer) {
@@ -496,9 +511,30 @@ void FFTcode(void * parameter)
static bool isFirstRun = false;
#ifdef FFT_USE_SLIDING_WINDOW
static float oldSamples[samplesFFT_2] = {0.0f}; // previous 50% of samples
static float* oldSamples = nullptr; // previous 50% of samples
static bool haveOldSamples = false; // for sliding window FFT
bool usingOldSamples = false;
if (!oldSamples) oldSamples = (float*) calloc(sizeof(float), samplesFFT_2); // allocate on first run
if (!oldSamples) { disableSoundProcessing = true; return; } // no memory -> die
#endif
bool success = true;
if ((vReal == nullptr) || (vImag == nullptr)) success = alocateFFTBuffers(); // allocate sample buffers on first run
if (success == false) { disableSoundProcessing = true; return; } // no memory -> die
// create FFT object - we have to do if after allocating buffers
#if defined(FFT_LIB_REV) && FFT_LIB_REV > 0x19
// arduinoFFT 2.x has a slightly different API
static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, true);
#else
// recommended version optimized by @softhack007 (API version 1.9)
#if defined(WLED_ENABLE_HUB75MATRIX) && defined(CONFIG_IDF_TARGET_ESP32)
static float* windowWeighingFactors = nullptr;
if (!windowWeighingFactors) windowWeighingFactors = (float*) calloc(sizeof(float), samplesFFT); // cache for FFT windowing factors - use heap
#else
static float windowWeighingFactors[samplesFFT] = {0.0f}; // cache for FFT windowing factors - use global RAM
#endif
static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, windowWeighingFactors);
#endif
#ifdef FFT_MAJORPEAK_HUMAN_EAR
@@ -542,7 +578,7 @@ void FFTcode(void * parameter)
#endif
// get a fresh batch of samples from I2S
memset(vReal, 0, sizeof(vReal)); // start clean
memset(vReal, 0, sizeof(float) * samplesFFT); // start clean
#ifdef FFT_USE_SLIDING_WINDOW
uint16_t readOffset;
if (haveOldSamples && (doSlidingFFT > 0)) {
@@ -635,7 +671,7 @@ void FFTcode(void * parameter)
#endif
// set imaginary parts to 0
memset(vImag, 0, sizeof(vImag));
memset(vImag, 0, sizeof(float) * samplesFFT);
#ifdef FFT_USE_SLIDING_WINDOW
memcpy(oldSamples, vReal+samplesFFT_2, sizeof(float) * samplesFFT_2); // copy last 50% to buffer (for sliding window FFT)
@@ -762,14 +798,14 @@ void FFTcode(void * parameter)
FFT_MajPeakSmth = FFT_MajPeakSmth + 0.42 * (FFT_MajorPeak - FFT_MajPeakSmth); // I like this "swooping peak" look
} else { // skip second run --> clear fft results, keep peaks
memset(vReal, 0, sizeof(vReal));
memset(vReal, 0, sizeof(float) * samplesFFT);
}
#if defined(WLED_DEBUG) || defined(SR_DEBUG) || defined(SR_STATS)
haveDoneFFT = true;
#endif
} else { // noise gate closed - only clear results as FFT was skipped. MIC samples are still valid when we do this.
memset(vReal, 0, sizeof(vReal));
memset(vReal, 0, sizeof(float) * samplesFFT);
FFT_MajorPeak = 1;
FFT_Magnitude = 0.001;
}

11
usermods/audioreactive/audio_source.h
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@@ -37,7 +37,8 @@
#define SRate_t int
#endif
constexpr i2s_port_t AR_I2S_PORT = I2S_NUM_0; // I2S port to use (do not change ! I2S_NUM_1 possible but this has limitation -> no MCLK routing, no ADC support)
constexpr i2s_port_t AR_I2S_PORT = I2S_NUM_0; // I2S port to use (do not change! I2S_NUM_1 possible but this has
// strong limitations -> no MCLK routing, no ADC support, no PDM support
//#include <driver/i2s_std.h>
//#include <driver/i2s_pdm.h>
@@ -73,6 +74,11 @@ constexpr i2s_port_t AR_I2S_PORT = I2S_NUM_0; // I2S port to use (do not c
// data type requested from the I2S driver - currently we always use 32bit
//#define I2S_USE_16BIT_SAMPLES // (experimental) define this to request 16bit - more efficient but possibly less compatible
#if defined(WLED_ENABLE_HUB75MATRIX) && defined(CONFIG_IDF_TARGET_ESP32)
// this is bitter, but necessary to survive
#define I2S_USE_16BIT_SAMPLES
#endif
#ifdef I2S_USE_16BIT_SAMPLES
#define I2S_SAMPLE_RESOLUTION I2S_BITS_PER_SAMPLE_16BIT
#define I2S_datatype int16_t
@@ -301,6 +307,9 @@ class I2SSource : public AudioSource {
#if defined(ARDUINO_ARCH_ESP32) && !defined(CONFIG_IDF_TARGET_ESP32S3) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
if (ESP.getChipRevision() == 0) _config.use_apll = false; // APLL is broken on ESP32 revision 0
#endif
#if defined(WLED_ENABLE_HUB75MATRIX)
_config.use_apll = false; // APLL needed for HUB75 DMA driver ?
#endif
#endif
if (_i2sMaster == false) {

10
wled00/ws.cpp
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@@ -186,6 +186,12 @@ static bool sendLiveLedsWs(uint32_t wsClient) // WLEDMM added "static"
AsyncWebSocketClient * wsc = ws.client(wsClient);
if (!wsc || wsc->queueLength() > 0) return false; //only send if queue free
#ifdef ARDUINO_ARCH_ESP32
static unsigned long ws_delay = 0;
if ((ws_delay > 0) && (millis() - ws_delay < 6000)) return false; // out of memory -> suspend for 6 seconds
else ws_delay = 0;
#endif
#ifdef ESP8266
constexpr size_t MAX_LIVE_LEDS_WS = 256U;
#else
@@ -223,6 +229,10 @@ static bool sendLiveLedsWs(uint32_t wsClient) // WLEDMM added "static"
last_err_time = millis();
}
errorFlag = ERR_LOW_WS_MEM;
#ifdef ARDUINO_ARCH_ESP32
ws_delay = millis(); // suspend for next 6 seconds
USER_PRINTLN("out of memory - live preview suspended for 6 seconds.");
#endif
return false; //out of memory
}
uint8_t* buffer = reinterpret_cast<uint8_t*>(wsBuf.data());