AudioSource improvements (work in progress)

-new methods: getType(), isInitailized(), postProcessSample()
- allow users to compile for RIGHT audio channel (-D I2S_USE_RIGHT_CHANNEL)
- better handling in case audio input driver failed to initialize
- removed some unneeded code and unneeded parameters

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."));
+          DEBUGSR_PRINT(F("AR: Generic I2S Microphone - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT));
-          audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF);
+          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."));
+          DEBUGSR_PRINTLN(F("AR: ES7243 Microphone (right channel only)."));
-          audioSource = new ES7243(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF);
+          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"));
+          DEBUGSR_PRINT(F("AR: SPH0645 Microphone - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT));
-          audioSource = new SPH0654(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF);
+          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"));
+          DEBUGSR_PRINT(F("AR: Generic I2S Microphone with Master Clock - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT));
-          audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF);
+          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"));
+          DEBUGSR_PRINT(F("AR: I2S PDM Microphone - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT));
-          audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 0, 0xFFFFFFFF);
+          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."));
+          DEBUGSR_PRINTLN(F("AR: Analog Microphone (left channel only)."));
-          // we don't do the down-shift by 16bit any more
+          audioSource = new I2SAdcSource(SAMPLE_RATE, BLOCK_SIZE);
-          //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);
           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 (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();
       }
 

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) :
+    I2SSource(int sampleRate, int blockSize) :
-      AudioSource(sampleRate, blockSize, lshift, mask) {
+      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
+          newSamples[i] = postProcessSample(newSamples[i]);  // perform postprocessing (needed for ADC samples)
-#ifdef I2S_SAMPLE_DOWNSCALE_TO_16BIT
+
-          if (_shift != 0)
-              newSamples[i] >>= 16;
-#endif
           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) :
+    ES7243(int sampleRate, int blockSize) :
-      I2SSource(sampleRate, blockSize, lshift, mask) {
+      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) :
+    I2SAdcSource(int sampleRate, int blockSize) :
-      I2SSource(sampleRate, blockSize, lshift, mask) {
+      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) :
+    SPH0654(int sampleRate, int blockSize) :
-      I2SSource(sampleRate, blockSize, lshift, mask)
+      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) {