|
- /* 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
|