Add envelope

AudioComputeEnvelope_F32.h

@@ -0,0 +1,118 @@
+/*
+ * AudioComputeEnvelope_F32
+ * 
+ * Created: Chip Audette, Feb 2017
+ * Purpose: This module extracts the envelope of the audio signal.
+ * Derived From: Core envelope extraction algorithm is from "smooth_env"
+ *     WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro
+ *     As of Feb 2017, CHAPRO license is listed as "Creative Commons?"
+ *          
+ * This processes a single stream fo audio data (ie, it is mono)       
+ *          
+ * MIT License.  use at your own risk.
+*/
+
+#ifndef _AudioComputeEnvelope_F32_h
+#define _AudioComputeEnvelope_F32_h
+
+#include <arm_math.h> //ARM DSP extensions.  for speed!
+#include <AudioStream_F32.h>
+
+class AudioComputeEnvelope_F32 : public AudioStream_F32
+{
+  //GUI: inputs:1, outputs:1  //this line used for automatic generation of GUI node
+  //GUI: shortName:envelope
+  public:
+    //default constructor
+    AudioComputeEnvelope_F32(void) : AudioStream_F32(1, inputQueueArray_f32),
+		sample_rate_Hz(AUDIO_SAMPLE_RATE) { setDefaultValues(); };
+	AudioComputeEnvelope_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32),
+		sample_rate_Hz(settings.sample_rate_Hz) { setDefaultValues(); };
+
+    //here's the method that does all the work
+    void update(void) {
+		
+		//get the input audio data block
+		audio_block_f32_t *in_block = AudioStream_F32::receiveReadOnly_f32();
+		if (!in_block) return;
+		
+		//check format
+		if (in_block->fs_Hz != sample_rate_Hz) {
+			Serial.println("AudioComputeEnvelope_F32: *** WARNING ***: Data sample rate does not match expected.");
+			Serial.println("AudioComputeEnvelope_F32: Changing sample rate.");
+			setSampleRate_Hz(in_block->fs_Hz);
+		}
+
+		//prepare an output data block
+		audio_block_f32_t *out_block = AudioStream_F32::allocate_f32();
+		if (!out_block) return;
+		
+		// //////////////////////add your processing here!
+		smooth_env(in_block->data, out_block->data, in_block->length);
+		out_block->length = in_block->length; out_block->fs_Hz = in_block->fs_Hz;
+		
+		//transmit the block and be done
+		AudioStream_F32::transmit(out_block);
+		AudioStream_F32::release(out_block);
+		AudioStream_F32::release(in_block);
+		
+    }
+	
+	//compute the smoothed signal envelope
+	//compute the envelope of the signal, not of the signal power)
+	void smooth_env(float x[], float y[], const int n) {
+        float  xab, xpk;
+        int k;
+    
+        // find envelope of x and return as y
+        //xpk = *ppk;                     // start with previous xpk
+		xpk = state_ppk;
+        for (k = 0; k < n; k++) {
+          xab = (x[k] >= 0.0f) ? x[k] : -x[k];
+          if (xab >= xpk) {
+              xpk = alfa * xpk + (1.f-alfa) * xab;
+          } else {
+              xpk = beta * xpk;
+          }
+          y[k] = xpk;
+        }
+        //*ppk = xpk;                     // save xpk for next time
+		state_ppk = xpk;
+	}
+	
+	//convert time constants from seconds to unitless parameters, from CHAPRO, agc_prepare.c
+	void setAttackRelease_msec(const float atk_msec, const float rel_msec) {
+		given_attack_msec = atk_msec;
+		given_release_msec = rel_msec;
+		
+		// convert ANSI attack & release times to filter time constants
+        float ansi_atk = 0.001f * atk_msec * sample_rate_Hz / 2.425f; 
+        float ansi_rel = 0.001f * rel_msec * sample_rate_Hz / 1.782f; 
+        alfa = (float) (ansi_atk / (1.0f + ansi_atk));
+        beta = (float) (ansi_rel / (10.f + ansi_rel));
+	}
+
+	void setDefaultValues(void) {
+		float32_t attack_msec = 5.0f;
+		float32_t release_msec = 50.0f;
+		setAttackRelease_msec(attack_msec, release_msec);
+		state_ppk = 0; //initialize
+	}
+
+    void setSampleRate_Hz(const float &fs_Hz) {
+		//change params that follow sample rate
+		
+		sample_rate_Hz = fs_Hz;
+	}
+	
+	void resetStates(void) { state_ppk = 1.0; }
+	
+  private:
+    audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
+	float32_t sample_rate_Hz;
+	float32_t given_attack_msec, given_release_msec;
+	float32_t alfa, beta;  //time constants, but in terms of samples, not seconds
+	float32_t state_ppk = 1.0f;
+};
+
+#endif

OpenAudio_ArduinoLibrary.h

@@ -2,6 +2,7 @@
 #include <AudioStream_F32.h>
 //include <AudioControlSGTL5000_Extended.h>
 #include <control_tlv320aic3206.h>
+#include "AudioComputeEnvelope_F32.h"
 #include <AudioConvert_F32.h>
 #include "AudioEffectEmpty_F32.h"
 #include <AudioEffectGain_F32.h>