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2D Waverly audio reactive.

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This commit is contained in:
Blaz Kristan
2022-06-12 22:17:17 +02:00
parent 922a3631ae
commit cc995ecef8
2 changed files with 76 additions and 62 deletions
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45
usermods/audioreactive/audio_reactive.h
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@@ -101,7 +101,7 @@ const uint16_t samples = 512; // This value MUST ALWAYS be a p
//unsigned int sampling_period_us;
//unsigned long microseconds;
static AudioSource *audioSource;
static AudioSource *audioSource = nullptr;
static byte soundSquelch = 10; // default squelch value for volume reactive routines
static byte sampleGain = 1; // default sample gain
@@ -162,7 +162,7 @@ void FFTcode(void * parameter) {
// Only run the FFT computing code if we're not in Receive mode
if (audioSyncEnabled & 0x02) continue;
audioSource->getSamples(vReal, samplesFFT);
if (audioSource) audioSource->getSamples(vReal, samplesFFT);
// old code - Last sample in vReal is our current mic sample
//micDataSm = (uint16_t)vReal[samples - 1]; // will do a this a bit later
@@ -739,7 +739,7 @@ class AudioReactive : public Usermod {
// usermod exchangeable data
// we will assign all usermod exportable data here as pointers to original variables or arrays and allocate memory for pointers
um_data = new um_data_t;
um_data->u_size = 8;
um_data->u_size = 11;
um_data->u_type = new um_types_t[um_data->u_size];
um_data->u_data = new void*[um_data->u_size];
um_data->u_data[0] = &maxVol;
@@ -758,25 +758,20 @@ class AudioReactive : public Usermod {
um_data->u_type[6] = UMT_DOUBLE;
um_data->u_data[7] = &FFT_Magnitude;
um_data->u_type[7] = UMT_DOUBLE;
um_data->u_data[8] = &sampleAvg;
um_data->u_type[8] = UMT_FLOAT;
um_data->u_data[9] = &soundAgc;
um_data->u_type[9] = UMT_BYTE;
um_data->u_data[10] = &sampleAgc;
um_data->u_type[10] = UMT_FLOAT;
//...
// these are values used by effects in soundreactive fork
//uint8_t *fftResult = um_data->;
//float *fftAvg = um_data->;
//float *fftBin = um_data->;
//float *fftCalc = um_data->;
//double FFT_MajorPeak = um_data->;
//double FFT_Magnitude = um_data->;
//float sampleAgc = um_data->;
//float sampleReal = um_data->;
//float multAgc = um_data->;
//float sampleAvg = um_data->;
//int16_t sample = um_data->;
//int16_t rawSampleAgc = um_data->;
//bool samplePeak = um_data->;
//uint8_t squelch = um_data->;
//uint8_t soundSquelch = um_data->;
//uint8_t soundAgc = um_data->;
//uint8_t maxVol = um_data->;
//uint8_t binNum = um_data->;
//uint16_t *myVals = um_data->;
//int16_t sample = um_data->;
@@ -788,43 +783,43 @@ class AudioReactive : public Usermod {
delay(100); // Give that poor microphone some time to setup.
switch (dmType) {
case 1:
DEBUGSR_PRINTLN("AS: Generic I2S Microphone.");
DEBUGSR_PRINTLN(F("AS: Generic I2S Microphone."));
audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF);
delay(100);
audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin);
if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin);
break;
case 2:
DEBUGSR_PRINTLN("AS: ES7243 Microphone.");
DEBUGSR_PRINTLN(F("AS: ES7243 Microphone."));
audioSource = new ES7243(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF);
delay(100);
audioSource->initialize(sdaPin, sclPin, i2swsPin, i2ssdPin, i2sckPin);
if (audioSource) audioSource->initialize(sdaPin, sclPin, i2swsPin, i2ssdPin, i2sckPin);
break;
case 3:
DEBUGSR_PRINTLN("AS: SPH0645 Microphone");
DEBUGSR_PRINTLN(F("AS: SPH0645 Microphone"));
audioSource = new SPH0654(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF);
delay(100);
audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin);
break;
case 4:
DEBUGSR_PRINTLN("AS: Generic I2S Microphone with Master Clock");
DEBUGSR_PRINTLN(F("AS: Generic I2S Microphone with Master Clock"));
audioSource = new I2SSourceWithMasterClock(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF);
delay(100);
audioSource->initialize(mclkPin, i2swsPin, i2ssdPin, i2sckPin);
if (audioSource) audioSource->initialize(mclkPin, i2swsPin, i2ssdPin, i2sckPin);
break;
case 5:
DEBUGSR_PRINTLN("AS: I2S PDM Microphone");
DEBUGSR_PRINTLN(F("AS: I2S PDM Microphone"));
audioSource = new I2SPdmSource(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF);
delay(100);
audioSource->initialize(i2swsPin, i2ssdPin);
if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin);
break;
case 0:
default:
DEBUGSR_PRINTLN("AS: Analog Microphone.");
DEBUGSR_PRINTLN(F("AS: Analog Microphone."));
// we don't do the down-shift by 16bit any more
//audioSource = new I2SAdcSource(SAMPLE_RATE, BLOCK_SIZE, -4, 0x0FFF); // request upscaling to 16bit - still produces too much noise
audioSource = new I2SAdcSource(SAMPLE_RATE, BLOCK_SIZE, 0, 0x0FFF); // keep at 12bit - less noise
delay(100);
audioSource->initialize(audioPin);
if (audioSource) audioSource->initialize(audioPin);
break;
}