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- /* Audio Library Guitar and 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.
- */
-
- #include "AudioTuner.h"
-
- #define HALF_BUFFER NUM_SAMPLES / 2
- #define QUARTER_BUFFER NUM_SAMPLES / 4
- #define EIGTH_BUFFER NUM_SAMPLES / 8
- #define SIXTEENTH_BUFFER NUM_SAMPLES / 16
- #define SAMPLE_RATE AUDIO_SAMPLE_RATE_EXACT / SAMPLE_SKIP
-
- #define LOOP1(a) a
- #define LOOP2(a) a LOOP1(a)
- #define LOOP3(a) a LOOP2(a)
- #define LOOP8(a) a LOOP3(a) a LOOP3(a)
- #define UNROLL(n,a) LOOP##n(a)
-
- #define WINDOW SAMPLE_SKIP - 1
-
- /**
- * audio update function.
- */
- void AudioTuner::update( void ) {
- audio_block_t *block;
- const int16_t *p, *end;
- block = receiveReadOnly( );
- if ( !block ) return;
- if ( !enabled ) {
- release( block );
- return;
- }
-
- p = block->data;
- end = p + AUDIO_BLOCK_SAMPLES;
-
- uint16_t *dst;
- if ( next_buffer ) dst = ( uint16_t * )buffer;
- else dst = ( uint16_t * )buffer + NUM_SAMPLES;
-
- uint8_t get_sample = 0;
- uint16_t count = block_count;
- do {
- if ( get_sample++ >= WINDOW ) {
- *( dst+count++ ) = *( uint16_t * )p;
- get_sample = 0;
- }
- } while ( p++ < end );
- release( block );
-
- if ( count >= NUM_SAMPLES ) {
- //digitalWriteFast(2, !digitalReadFast(2));
- next_buffer = !next_buffer;
- process_buffer = true;
- tau_global = 1;
- yin_idx = 1;
- running_sum = 0;
- count = 0;
- }
- block_count = count;
-
- if ( process_buffer ) {
- //digitalWriteFast(0, HIGH);
- uint16_t tau;
- uint16_t next;
- next = next_buffer;
- tau = tau_global;
- do {
- int64_t sum = 0;
- const int16_t *end, *buf;
- if ( next ) buf = buffer + NUM_SAMPLES;
- else buf = buffer;
- end = buf + HALF_BUFFER;
- do {
- int16_t current, lag, delta;
- UNROLL( 8,
- lag = *( buf + tau );
- current = *buf++;
- delta = current - lag;
- sum += delta*delta;
- );
- } while ( buf < end );
-
- running_sum += sum;
- yin_buffer[yin_idx] = sum*tau;
- rs_buffer[yin_idx] = running_sum;
- yin_idx = ( ++yin_idx >= 5 ) ? 0 : yin_idx;
- tau = estimate( yin_buffer, rs_buffer, yin_idx, tau );
- if ( tau == 0 ) {
- process_buffer = false;
- new_output = true;
- //digitalWriteFast(0, LOW);
- return;
- }
- else if ( tau >= HALF_BUFFER ) {
- process_buffer = false;
- new_output = false;
- //digitalWriteFast(0, LOW);
- return;
- }
- } while ( tau <= ( tau_global + 31 ) );
- tau_global = tau;
- //digitalWriteFast(0, LOW);
- }
- }
-
- /**
- * check the sampled data for fundmental 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 AudioTuner::estimate( int64_t *yin, int64_t *rs, uint16_t head, uint16_t tau ) {
- const int64_t *p = ( int64_t * )yin;
- const int64_t *r = ( int64_t * )rs;
- uint16_t period = 0, _tau, _head;
- _tau = tau;
- _head = head;
-
- if ( _tau > 4 ) {
- uint16_t idx0, idx1, idx2;
- idx0 = _head;
- idx1 = _head + 1;
- idx1 = ( idx1 >= 5 ) ? 0 : idx1;
- idx2 = head + 2;
- idx2 = ( idx2 >= 5 ) ? 0 : idx2;
-
- float s0, s1, s2;
- s0 = ( ( float )*( p+idx0 ) / r[idx0] );
- s1 = ( ( float )*( p+idx1 ) / r[idx1] );
- s2 = ( ( float )*( p+idx2 ) / r[idx2] );
-
- if ( s1 < threshold && s1 < s2 ) {
- period = _tau - 3;
- periodicity = 1 - s1;
- data = period + 0.5f * ( s0 - s2 ) / ( s0 - 2.0f * s1 + s2 );
- return 0;
- }
-
- if ( s1 > 2.2 ) return _tau + 2;
- else return _tau + 1;
- }
- return _tau + 1;
- }
-
- /**
- * available
- *
- * @return true if data is ready else false
- */
- bool AudioTuner::available( void ) {
- __disable_irq();
- bool flag = new_output;
- if (flag) new_output = false;
- __enable_irq();
- return flag;
- }
-
- /**
- * read processes the data samples for the Yin algorithm.
- *
- * @return frequency in hertz
- */
- float AudioTuner::read( void ) {
- return SAMPLE_RATE / data;
- }
- /**
- * Periodicity of the sampled signal from Yin algorithm from read function.
- *
- * @return periodicity
- */
- float AudioTuner::probability( void ) {
- return periodicity;
- }
- /**
- * Initialise parameters.
- *
- * @param thresh Allowed uncertainty
- */
- void AudioTuner::set_params( float thresh ) {
- __disable_irq( );
- //arm_cfft_radix4_init_q15(&fft_inst, 1024, 0, 1);
- //arm_cfft_radix4_init_q15(&ifft_inst, 1024, 1, 1);
- threshold = thresh;
- periodicity = 0.0f;
- block_count = 0;
- enabled = true;
- __enable_irq( );
- }
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