teensy-audio-tuner/include/audio/AudioTuner.h

/* Audio Library Note Frequency Detection & Guitar/Bass Tuner
 * Copyright (c) 2015, Colin Duffy
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice, development funding notice, and this permission
 * notice shall be included in all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
 * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */

#ifndef AudioTuner_h_
#define AudioTuner_h_

#include <core/core_pins.h>
#include <core/AudioStream.h>
#include <arm_math.h>
/***********************************************************************
 *              Safe to adjust these values below                      *
 *                                                                     *
 *  This parameter defines the size of the buffer.                     *
 *                                                                     *
 *  1.  AUDIO_GUITARTUNER_BLOCKS -  Buffer size is 128 * AUDIO_BLOCKS. *
 *                      The more AUDIO_GUITARTUNER_BLOCKS the lower    *
 *                      the frequency you can detect. The default      *
 *                      (24) is set to measure down to 29.14 Hz        *
 *                      or B(flat)0.                                   *
 *                                                                     *
 *  2.  MAX_COEFF - Maxium number of coefficeints for the FIR filter.  *
 *                                                                     *
 ***********************************************************************/
#define AUDIO_GUITARTUNER_BLOCKS  24
#define MAX_COEFF                 200
/***********************************************************************/

class AudioTuner : public AudioStream {
public:
    /**
     *  constructor to setup Audio Library and initialize
     *
     *  @return none
     */
  AudioTuner(void)
	  : AudioStream(1, inputQueueArray)
	  , data(0.0)
	  , coeff_p(NULL)
	  , coeff_size(0)
	  , new_output(false)
	  , enabled(false)

  {}

  /**
   *  initialize variables and start
   *
   *  @param threshold Allowed uncertainty
   *  @param coeff     coefficients for fir filter
   *  @param taps      number of coefficients, even
   *  @param factor    must be power of 2
   */
  void begin(float threshold, int16_t *coeff, uint8_t taps, uint8_t factor);

  /**
   *  sets threshold value
   *
   *  @param thresh
   *  @return none
   */
  void threshold(float p);

  /**
   *  triggers true when valid frequency is found
   *
   *  @return flag to indicate valid frequency is found
   */
  bool available(void);
  /**
   *  get frequency
   *
   *  @return frequency in hertz
   */
  float read(void);

  /**
   *  get predicitity
   *
   *  @return probability of frequency found
   */
  float probability(void);

  /**
   *  fir decimation coefficents
   *
   *  @return none
   */
  void coeff(int16_t *p, int n);

  /**
   *  disable yin
   *
   *  @return none
   */
  void disable(void);

  /**
   *  Audio Library calls this update function ~2.9ms
   *
   *  @return none
   */
  virtual void update(void);

private:
    /**
     *  check the sampled data for fundamental frequency
     *
     *  @param yin  buffer to hold sum*tau value
     *  @param rs   buffer to hold running sum for sampled window
     *  @param head buffer index
     *  @param tau  lag we are currently working on this gets incremented
     *
     *  @return tau
     */
    uint16_t estimate( uint64_t *yin, uint64_t *rs, uint16_t head, uint16_t tau );

    /**
     *  process audio data
     *
     *  @return none
     */
    void process( int16_t *p );

    /**
     *  Variables
     */
    float    periodicity, yin_threshold, data;
    uint64_t running_sum, yin_buffer[5], rs_buffer[5];
    uint16_t tau_global;
    int16_t  AudioBuffer[AUDIO_GUITARTUNER_BLOCKS*AUDIO_BLOCK_SAMPLES] __attribute__ ( ( aligned ( 4 ) ) );
    int16_t  coeff_state[AUDIO_BLOCK_SAMPLES + MAX_COEFF];
    int16_t  *coeff_p;
    uint8_t  yin_idx, state, coeff_size, decimation_factor, decimation_shift;
    volatile bool new_output, process_buffer, enabled;
    audio_block_t *inputQueueArray[1];
    arm_fir_decimate_instance_q15 firDecimateInst;
};
#endif