diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 35606324..fb8cafb3 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -728,11 +728,11 @@ class AudioReactive : public Usermod { } - void receiveAudioData() + bool receiveAudioData() // check & process new data. return TRUE in case that new audio data was received. { - if (!udpSyncConnected) return; + if (!udpSyncConnected) return false; //DEBUGSR_PRINTLN("Checking for UDP Microphone Packet"); - + bool haveFreshData = false; size_t packetSize = fftUdp.parsePacket(); if (packetSize > 5) { //DEBUGSR_PRINTLN("Received UDP Sync Packet"); @@ -774,8 +774,10 @@ class AudioReactive : public Usermod { FFT_Magnitude = my_magnitude; FFT_MajorPeak = receivedPacket->FFT_MajorPeak; //DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet"); + haveFreshData = true; } } + return haveFreshData; } @@ -822,50 +824,54 @@ class AudioReactive : public Usermod { delay(100); // Give that poor microphone some time to setup. switch (dmType) { case 1: - DEBUGSR_PRINTLN(F("AS: Generic I2S Microphone.")); - audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF); + DEBUGSR_PRINT(F("AR: Generic I2S Microphone - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT)); + audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE); delay(100); if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin); break; case 2: - DEBUGSR_PRINTLN(F("AS: ES7243 Microphone.")); - audioSource = new ES7243(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF); + DEBUGSR_PRINTLN(F("AR: ES7243 Microphone (right channel only).")); + audioSource = new ES7243(SAMPLE_RATE, BLOCK_SIZE); delay(100); if (audioSource) audioSource->initialize(sdaPin, sclPin, i2swsPin, i2ssdPin, i2sckPin, mclkPin); break; case 3: - DEBUGSR_PRINTLN(F("AS: SPH0645 Microphone")); - audioSource = new SPH0654(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF); + DEBUGSR_PRINT(F("AR: SPH0645 Microphone - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT)); + audioSource = new SPH0654(SAMPLE_RATE, BLOCK_SIZE); delay(100); audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin); break; case 4: - DEBUGSR_PRINTLN(F("AS: Generic I2S Microphone with Master Clock")); - audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF); + DEBUGSR_PRINT(F("AR: Generic I2S Microphone with Master Clock - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT)); + audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE); delay(100); if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin, mclkPin); break; case 5: - DEBUGSR_PRINTLN(F("AS: I2S PDM Microphone")); - audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF); + DEBUGSR_PRINT(F("AR: I2S PDM Microphone - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT)); + audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE); delay(100); if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin); break; case 0: default: - 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 + DEBUGSR_PRINTLN(F("AR: Analog Microphone (left channel only).")); + audioSource = new I2SAdcSource(SAMPLE_RATE, BLOCK_SIZE); delay(100); if (audioSource) audioSource->initialize(audioPin); break; } - delay(250); // give mictophone enough time to initialise + delay(250); // give microphone enough time to initialise - if (!audioSource) enabled = false; // audio failed to initialise - if (enabled) onUpdateBegin(false); // create FFT task - if (enabled) disableSoundProcessing = false; + if (!audioSource) enabled = false; // audio failed to initialise + if (enabled) onUpdateBegin(false); // create FFT task + if (FFT_Task == nullptr) enabled = false; // FFT task creation failed + if (enabled) disableSoundProcessing = false; // all good - enable audio processing + + if((!audioSource) || (!audioSource->isInitialized())) { // audio source failed to initialize. Still stay "enabled", as there might be input arriving via UDP Sound Sync + DEBUGSR_PRINTLN(F("AR: Failed to initialize sound input driver. Please check input PIN settings.")); + disableSoundProcessing = true; + } initDone = true; } @@ -1019,7 +1025,7 @@ class AudioReactive : public Usermod { // Begin UDP Microphone Sync if ((audioSyncEnabled & 0x02) && millis() - lastTime > delayMs) { // Only run the audio listener code if we're in Receive mode - receiveAudioData(); + (void) receiveAudioData(); // ToDo: use return value for something meaningfull lastTime = millis(); } diff --git a/usermods/audioreactive/audio_source.h b/usermods/audioreactive/audio_source.h index 168b3aca..dc46f15f 100644 --- a/usermods/audioreactive/audio_source.h +++ b/usermods/audioreactive/audio_source.h @@ -18,19 +18,36 @@ Until this configuration is moved to the webinterface */ +// if you have problems to get your microphone work on the left channel, uncomment the following line +//#define I2S_USE_RIGHT_CHANNEL // (experimental) define this to use right channel (digital mics only) +#ifdef I2S_USE_RIGHT_CHANNEL +#define I2S_MIC_CHANNEL I2S_CHANNEL_FMT_ONLY_RIGHT +#define I2S_MIC_CHANNEL_TEXT "right channel only." +#else +#define I2S_MIC_CHANNEL I2S_CHANNEL_FMT_ONLY_LEFT +#define I2S_MIC_CHANNEL_TEXT "left channel only." +#endif + +// Uncomment the line below to utilize ADC1 _exclusively_ for I2S sound input. +// benefit: analog mic inputs will be sampled contiously -> better response times and less "glitches" +// WARNING: this option WILL lock-up your device in case that any other analogRead() operation is performed; +// for example if you want to read "analog buttons" +//#define I2S_GRAB_ADC1_COMPLETELY // (experimental) continously sample analog ADC microphone. WARNING will cause analogRead() lock-up + // 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 #ifdef I2S_USE_16BIT_SAMPLES #define I2S_SAMPLE_RESOLUTION I2S_BITS_PER_SAMPLE_16BIT #define I2S_datatype int16_t +#define I2S_unsigned_datatype uint16_t #undef I2S_SAMPLE_DOWNSCALE_TO_16BIT #else #define I2S_SAMPLE_RESOLUTION I2S_BITS_PER_SAMPLE_32BIT #define I2S_datatype int32_t +#define I2S_unsigned_datatype uint32_t #define I2S_SAMPLE_DOWNSCALE_TO_16BIT #endif - /* Interface class AudioSource serves as base class for all microphone types This enables accessing all microphones with one single interface @@ -65,15 +82,23 @@ class AudioSource { /* Get an up-to-date sample without DC offset */ virtual int getSampleWithoutDCOffset() { return _sampleNoDCOffset; }; + /* check if the audio source driver was initialized successfully */ + virtual bool isInitialized(void) {return(_initialized);} + + /* identify Audiosource type - I2S-ADC or I2S-digital */ + typedef enum{Type_unknown=0, Type_I2SAdc=1, Type_I2SDigital=2} AudioSourceType; + virtual AudioSourceType getType(void) {return(Type_I2SDigital);} // default is "I2S digital source" - ADC type overrides this method + protected: + /* Post-process audio sample - currently on needed for I2SAdcSource*/ + virtual I2S_datatype postProcessSample(I2S_datatype sample_in) {return(sample_in);} // default method can be overriden by instances (ADC) that need sample postprocessing + // Private constructor, to make sure it is not callable except from derived classes - AudioSource(int sampleRate, int blockSize, int16_t lshift, uint32_t mask) : + AudioSource(int sampleRate, int blockSize) : _sampleRate(sampleRate), _blockSize(blockSize), _sampleNoDCOffset(0), _dcOffset(0.0f), - _shift(lshift), - _mask(mask), _initialized(false) {}; @@ -81,8 +106,6 @@ class AudioSource { int _blockSize; // I2S block size volatile int _sampleNoDCOffset; // Up-to-date sample without DCOffset float _dcOffset; // Rolling average DC offset - int16_t _shift; // Shift obtained samples to the right (positive) or left(negative) by this amount - uint32_t _mask; // Bitmask for sample data after shifting. Bitmask 0X0FFF means that we need to convert 12bit ADC samples from unsigned to signed bool _initialized; // Gets set to true if initialization is successful }; @@ -91,13 +114,13 @@ class AudioSource { */ class I2SSource : public AudioSource { public: - I2SSource(int sampleRate, int blockSize, int16_t lshift, uint32_t mask) : - AudioSource(sampleRate, blockSize, lshift, mask) { + I2SSource(int sampleRate, int blockSize) : + AudioSource(sampleRate, blockSize) { _config = { .mode = i2s_mode_t(I2S_MODE_MASTER | I2S_MODE_RX), .sample_rate = _sampleRate, .bits_per_sample = I2S_SAMPLE_RESOLUTION, - .channel_format = I2S_CHANNEL_FMT_ONLY_LEFT, + .channel_format = I2S_MIC_CHANNEL, #if ESP_IDF_VERSION >= ESP_IDF_VERSION_VAL(4, 2, 0) .communication_format = i2s_comm_format_t(I2S_COMM_FORMAT_STAND_I2S), #else @@ -193,26 +216,16 @@ class I2SSource : public AudioSource { // Store samples in sample buffer and update DC offset for (int i = 0; i < num_samples; i++) { - // pre-shift samples down to 16bit -#ifdef I2S_SAMPLE_DOWNSCALE_TO_16BIT - if (_shift != 0) - newSamples[i] >>= 16; -#endif + newSamples[i] = postProcessSample(newSamples[i]); // perform postprocessing (needed for ADC samples) + float currSample = 0.0f; - if(_shift > 0) - currSample = (float) (newSamples[i] >> _shift); - else { - if(_shift < 0) - currSample = (float) (newSamples[i] << (- _shift)); // need to "pump up" 12bit ADC to full 16bit as delivered by other digital mics - else #ifdef I2S_SAMPLE_DOWNSCALE_TO_16BIT - currSample = (float) newSamples[i] / 65536.0f; // _shift == 0 -> use the chance to keep lower 16bits + currSample = (float) newSamples[i] / 65536.0f; // 32bit input -> 16bit; keeping lower 16bits as decimal places #else - currSample = (float) newSamples[i]; + currSample = (float) newSamples[i]; // 16bit input -> use as-is #endif - } buffer[i] = currSample; - _dcOffset = ((_dcOffset * 31) + currSample) / 32; + _dcOffset = ((_dcOffset * 31.0f) + currSample) / 32.0f; } // Update no-DC sample @@ -275,8 +288,8 @@ class ES7243 : public I2SSource { } public: - ES7243(int sampleRate, int blockSize, int16_t lshift, uint32_t mask) : - I2SSource(sampleRate, blockSize, lshift, mask) { + ES7243(int sampleRate, int blockSize) : + I2SSource(sampleRate, blockSize) { _config.channel_format = I2S_CHANNEL_FMT_ONLY_RIGHT; }; @@ -314,8 +327,8 @@ public: */ class I2SAdcSource : public I2SSource { public: - I2SAdcSource(int sampleRate, int blockSize, int16_t lshift, uint32_t mask) : - I2SSource(sampleRate, blockSize, lshift, mask) { + I2SAdcSource(int sampleRate, int blockSize) : + I2SSource(sampleRate, blockSize) { _config = { .mode = i2s_mode_t(I2S_MODE_MASTER | I2S_MODE_RX | I2S_MODE_ADC_BUILT_IN), .sample_rate = _sampleRate, @@ -332,6 +345,9 @@ class I2SAdcSource : public I2SSource { }; } + /* identify Audiosource type - I2S-ADC*/ + AudioSourceType getType(void) {return(Type_I2SAdc);} + void initialize(int8_t audioPin, int8_t = I2S_PIN_NO_CHANGE, int8_t = I2S_PIN_NO_CHANGE, int8_t = I2S_PIN_NO_CHANGE, int8_t = I2S_PIN_NO_CHANGE, int8_t = I2S_PIN_NO_CHANGE) { if(!pinManager.allocatePin(audioPin, false, PinOwner::UM_Audioreactive)) { return; @@ -432,8 +448,8 @@ class I2SAdcSource : public I2SSource { */ class SPH0654 : public I2SSource { public: - SPH0654(int sampleRate, int blockSize, int16_t lshift, uint32_t mask) : - I2SSource(sampleRate, blockSize, lshift, mask) + SPH0654(int sampleRate, int blockSize) : + I2SSource(sampleRate, blockSize) {} void initialize(uint8_t i2swsPin, uint8_t i2ssdPin, uint8_t i2sckPin, int8_t = I2S_PIN_NO_CHANGE, int8_t = I2S_PIN_NO_CHANGE, int8_t = I2S_PIN_NO_CHANGE) {