Update AudioFilterFIR for audio_block_samples

AudioFilterFIR_F32.h

@@ -22,34 +22,34 @@
 class AudioFilterFIR_F32 : public AudioStream_F32
 {
 //GUI: inputs:1, outputs:1  //this line used for automatic generation of GUI node  
-public:
-  AudioFilterFIR_F32(void): AudioStream_F32(1,inputQueueArray), coeff_p(FIR_F32_PASSTHRU) {
-  }
-  void begin(const float32_t *cp, int n_coeffs) {
+	public:
+		AudioFilterFIR_F32(void): AudioStream_F32(1,inputQueueArray), 
+			coeff_p(FIR_F32_PASSTHRU), n_coeffs(1), configured_block_size(0) {	}
+
+		//initialize the FIR filter by giving it the filter coefficients
+		void begin(const float32_t *cp, const int _n_coeffs) { begin(cp, _n_coeffs, AUDIO_BLOCK_SAMPLES); } //assume that the block size is the maximum
+		void begin(const float32_t *cp, const int _n_coeffs, const int block_size) {  //or, you can provide it with the block size
 			coeff_p = cp;
+			n_coeffs = _n_coeffs;
+
 			// Initialize FIR instance (ARM DSP Math Library)
 			if (coeff_p && (coeff_p != FIR_F32_PASSTHRU) && n_coeffs <= FIR_MAX_COEFFS) {
-      arm_fir_init_f32(&fir_inst, n_coeffs, (float32_t *)coeff_p,  &StateF32[0], AUDIO_BLOCK_SAMPLES);
-      //if (arm_fir_init_f32(&fir_inst, n_coeffs, (float32_t *)coeff_p,  &StateF32[0], AUDIO_BLOCK_SAMPLES) != ARM_MATH_SUCCESS) {
-        // n_coeffs must be an even number, 4 or larger
-        //coeff_p = NULL;
-      //}
+				arm_fir_init_f32(&fir_inst, n_coeffs, (float32_t *)coeff_p,  &StateF32[0], block_size);
+				configured_block_size = block_size;
 			}
 		}
-  void end(void) {
-    coeff_p = NULL;
-  }
+		void end(void) {  coeff_p = NULL; }
 		virtual void update(void);
 
-  void setBlockDC(void) {
-    //helper function that sets this up for a first-order HP filter at 20Hz
+		//void setBlockDC(void) {}	//helper function that sets this up for a first-order HP filter at 20Hz
 		
-  }
-private:
+	private:
 		audio_block_f32_t *inputQueueArray[1];
 
 		// pointer to current coefficients or NULL or FIR_PASSTHRU
 		const float32_t *coeff_p;
+		int n_coeffs;
+		int configured_block_size;
 
 		// ARM DSP Math library filter instance
 		arm_fir_instance_f32 fir_inst;
@@ -81,6 +81,13 @@ void AudioFilterFIR_F32::update(void)
 	// get a block for the FIR output
 	b_new = AudioStream_F32::allocate_f32();
 	if (b_new) {
+		//check to make sure our FIR instance has the right size
+		if (block->length != configured_block_size) {
+			//doesn't match.  re-initialize
+			Serial.println("AudioFilterFIR_F32: block size doesn't match.  Re-initializing FIR.");
+			begin(coeff_p, n_coeffs, block->length);
+		}
+		//apply the FIR
 		arm_fir_f32(&fir_inst, (float32_t *)block->data, (float32_t *)b_new->data, block->length);
 		AudioStream_F32::transmit(b_new); // send the FIR output
 		AudioStream_F32::release(b_new);