268 lines
7.8 KiB
C++
Executable file
268 lines
7.8 KiB
C++
Executable file
/* Audio Library Note Frequency Detection & Guitar/Bass Tuner
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* Copyright (c) 2015, Colin Duffy
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice, development funding notice, and this permission
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* notice shall be included in all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#include <Arduino.h>
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#include "analyze_notefreq.h"
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#include "utility/dspinst.h"
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#include "arm_math.h"
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#define HALF_BLOCKS AUDIO_GUITARTUNER_BLOCKS * 64
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/**
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* Copy internal blocks of data to class buffer
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*
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* @param destination destination address
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* @param source source address
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*/
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static void copy_buffer(void *destination, const void *source) {
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const uint16_t *src = ( const uint16_t * )source;
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uint16_t *dst = ( uint16_t * )destination;
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for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) *dst++ = (*src++);
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}
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/**
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* Virtual function to override from Audio Library
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*/
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void AudioAnalyzeNoteFrequency::update( void ) {
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audio_block_t *block;
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block = receiveReadOnly();
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if (!block) return;
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if ( !enabled ) {
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release( block );
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return;
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}
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if ( next_buffer ) {
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blocklist1[state++] = block;
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if ( !first_run && process_buffer ) process( );
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} else {
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blocklist2[state++] = block;
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if ( !first_run && process_buffer ) process( );
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}
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if ( state >= AUDIO_GUITARTUNER_BLOCKS ) {
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if ( next_buffer ) {
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if ( !first_run && process_buffer ) process( );
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for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) copy_buffer( AudioBuffer+( i * 0x80 ), blocklist1[i]->data );
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for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) release( blocklist1[i] );
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next_buffer = false;
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} else {
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if ( !first_run && process_buffer ) process( );
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for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) copy_buffer( AudioBuffer+( i * 0x80 ), blocklist2[i]->data );
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for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) release( blocklist2[i] );
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next_buffer = true;
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}
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process_buffer = true;
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first_run = false;
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state = 0;
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}
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}
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/**
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* Start the Yin algorithm
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*
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* TODO: Significant speed up would be to use spectral domain to find fundamental frequency.
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* This paper explains: https://aubio.org/phd/thesis/brossier06thesis.pdf -> Section 3.2.4
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* page 79. Might have to downsample for low fundmental frequencies because of fft buffer
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* size limit.
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*/
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void AudioAnalyzeNoteFrequency::process( void ) {
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const int16_t *p;
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p = AudioBuffer;
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uint16_t cycles = 64;
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uint16_t tau = tau_global;
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do {
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uint16_t x = 0;
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uint64_t sum = 0;
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do {
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int16_t current, lag, delta;
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lag = *( ( int16_t * )p + ( x+tau ) );
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current = *( ( int16_t * )p+x );
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delta = ( current-lag );
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sum += delta * delta;
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x += 4;
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lag = *( ( int16_t * )p + ( x+tau ) );
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current = *( ( int16_t * )p+x );
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delta = ( current-lag );
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sum += delta * delta;
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x += 4;
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lag = *( ( int16_t * )p + ( x+tau ) );
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current = *( ( int16_t * )p+x );
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delta = ( current-lag );
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sum += delta * delta;
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x += 4;
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lag = *( ( int16_t * )p + ( x+tau ) );
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current = *( ( int16_t * )p+x );
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delta = ( current-lag );
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sum += delta * delta;
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x += 4;
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} while ( x < HALF_BLOCKS );
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uint64_t rs = running_sum;
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rs += sum;
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yin_buffer[yin_idx] = sum*tau;
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rs_buffer[yin_idx] = rs;
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running_sum = rs;
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yin_idx = ( ++yin_idx >= 5 ) ? 0 : yin_idx;
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tau = estimate( yin_buffer, rs_buffer, yin_idx, tau );
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if ( tau == 0 ) {
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process_buffer = false;
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new_output = true;
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yin_idx = 1;
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running_sum = 0;
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tau_global = 1;
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return;
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}
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} while ( --cycles );
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//digitalWriteFast(10, LOW);
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if ( tau >= HALF_BLOCKS ) {
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process_buffer = false;
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new_output = false;
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yin_idx = 1;
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running_sum = 0;
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tau_global = 1;
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return;
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}
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tau_global = tau;
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}
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/**
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* check the sampled data for fundamental frequency
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*
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* @param yin buffer to hold sum*tau value
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* @param rs buffer to hold running sum for sampled window
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* @param head buffer index
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* @param tau lag we are currently working on gets incremented
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*
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* @return tau
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*/
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uint16_t AudioAnalyzeNoteFrequency::estimate( uint64_t *yin, uint64_t *rs, uint16_t head, uint16_t tau ) {
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const uint64_t *y = ( uint64_t * )yin;
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const uint64_t *r = ( uint64_t * )rs;
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uint16_t _tau, _head;
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const float thresh = yin_threshold;
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_tau = tau;
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_head = head;
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if ( _tau > 4 ) {
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uint16_t idx0, idx1, idx2;
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idx0 = _head;
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idx1 = _head + 1;
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idx1 = ( idx1 >= 5 ) ? 0 : idx1;
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idx2 = head + 2;
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idx2 = ( idx2 >= 5 ) ? 0 : idx2;
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float s0, s1, s2;
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s0 = ( ( float )*( y+idx0 ) / *( r+idx0 ) );
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s1 = ( ( float )*( y+idx1 ) / *( r+idx1 ) );
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s2 = ( ( float )*( y+idx2 ) / *( r+idx2 ) );
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if ( s1 < thresh && s1 < s2 ) {
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uint16_t period = _tau - 3;
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periodicity = 1 - s1;
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data = period + 0.5f * ( s0 - s2 ) / ( s0 - 2.0f * s1 + s2 );
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return 0;
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}
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}
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return _tau + 1;
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}
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/**
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* Initialise
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*
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* @param threshold Allowed uncertainty
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*/
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void AudioAnalyzeNoteFrequency::begin( float threshold ) {
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__disable_irq( );
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process_buffer = false;
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yin_threshold = threshold;
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periodicity = 0.0f;
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next_buffer = true;
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running_sum = 0;
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tau_global = 1;
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first_run = true;
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yin_idx = 1;
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enabled = true;
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state = 0;
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data = 0.0f;
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__enable_irq( );
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}
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/**
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* available
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*
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* @return true if data is ready else false
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*/
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bool AudioAnalyzeNoteFrequency::available( void ) {
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__disable_irq( );
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bool flag = new_output;
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if ( flag ) new_output = false;
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__enable_irq( );
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return flag;
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}
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/**
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* read processes the data samples for the Yin algorithm.
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*
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* @return frequency in hertz
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*/
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float AudioAnalyzeNoteFrequency::read( void ) {
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__disable_irq( );
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float d = data;
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__enable_irq( );
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return AUDIO_SAMPLE_RATE_EXACT / d;
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}
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/**
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* Periodicity of the sampled signal from Yin algorithm from read function.
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*
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* @return periodicity
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*/
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float AudioAnalyzeNoteFrequency::probability( void ) {
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__disable_irq( );
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float p = periodicity;
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__enable_irq( );
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return p;
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}
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/**
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* Initialise parameters.
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*
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* @param thresh Allowed uncertainty
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*/
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void AudioAnalyzeNoteFrequency::threshold( float p ) {
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__disable_irq( );
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yin_threshold = p;
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__enable_irq( );
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}
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