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/* 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.
*/
#include <Arduino.h>
#include "analyze_notefreq.h"
#include "utility/dspinst.h"
#include "arm_math.h"
#define HALF_BLOCKS AUDIO_GUITARTUNER_BLOCKS * 64
/**
* Copy internal blocks of data to class buffer
*
* @param destination destination address
* @param source source address
*/
static void copy_buffer(void *destination, const void *source) {
const uint16_t *src = ( const uint16_t * )source;
uint16_t *dst = ( uint16_t * )destination;
for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) *dst++ = (*src++);
}
/**
* Virtual function to override from Audio Library
*/
void AudioAnalyzeNoteFrequency::update( void ) {
audio_block_t *block;
block = receiveReadOnly();
if (!block) return;
if ( !enabled ) {
release( block );
return;
}
if ( next_buffer ) {
blocklist1[state++] = block;
if ( !first_run && process_buffer ) process( );
} else {
blocklist2[state++] = block;
if ( !first_run && process_buffer ) process( );
}
if ( state >= AUDIO_GUITARTUNER_BLOCKS ) {
if ( next_buffer ) {
if ( !first_run && process_buffer ) process( );
for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) copy_buffer( AudioBuffer+( i * 0x80 ), blocklist1[i]->data );
for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) release( blocklist1[i] );
next_buffer = false;
} else {
if ( !first_run && process_buffer ) process( );
for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) copy_buffer( AudioBuffer+( i * 0x80 ), blocklist2[i]->data );
for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) release( blocklist2[i] );
next_buffer = true;
}
process_buffer = true;
first_run = false;
state = 0;
}
}
/**
* Start the Yin algorithm
*
* TODO: Significant speed up would be to use spectral domain to find fundamental frequency.
* This paper explains: https://aubio.org/phd/thesis/brossier06thesis.pdf -> Section 3.2.4
* page 79. Might have to downsample for low fundmental frequencies because of fft buffer
* size limit.
*/
void AudioAnalyzeNoteFrequency::process( void ) {
const int16_t *p;
p = AudioBuffer;
uint16_t cycles = 64;
uint16_t tau = tau_global;
do {
uint16_t x = 0;
uint64_t sum = 0;
do {
int16_t current, lag, delta;
lag = *( ( int16_t * )p + ( x+tau ) );
current = *( ( int16_t * )p+x );
delta = ( current-lag );
sum += delta * delta;
x += 4;
lag = *( ( int16_t * )p + ( x+tau ) );
current = *( ( int16_t * )p+x );
delta = ( current-lag );
sum += delta * delta;
x += 4;
lag = *( ( int16_t * )p + ( x+tau ) );
current = *( ( int16_t * )p+x );
delta = ( current-lag );
sum += delta * delta;
x += 4;
lag = *( ( int16_t * )p + ( x+tau ) );
current = *( ( int16_t * )p+x );
delta = ( current-lag );
sum += delta * delta;
x += 4;
} while ( x < HALF_BLOCKS );
uint64_t rs = running_sum;
rs += sum;
yin_buffer[yin_idx] = sum*tau;
rs_buffer[yin_idx] = rs;
running_sum = rs;
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;
yin_idx = 1;
running_sum = 0;
tau_global = 1;
return;
}
} while ( --cycles );
//digitalWriteFast(10, LOW);
if ( tau >= HALF_BLOCKS ) {
process_buffer = false;
new_output = false;
yin_idx = 1;
running_sum = 0;
tau_global = 1;
return;
}
tau_global = tau;
}
/**
* 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 gets incremented
*
* @return tau
*/
uint16_t AudioAnalyzeNoteFrequency::estimate( uint64_t *yin, uint64_t *rs, uint16_t head, uint16_t tau ) {
const uint64_t *y = ( uint64_t * )yin;
const uint64_t *r = ( uint64_t * )rs;
uint16_t _tau, _head;
const float thresh = yin_threshold;
_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 )*( y+idx0 ) / *( r+idx0 ) );
s1 = ( ( float )*( y+idx1 ) / *( r+idx1 ) );
s2 = ( ( float )*( y+idx2 ) / *( r+idx2 ) );
if ( s1 < thresh && s1 < s2 ) {
uint16_t period = _tau - 3;
periodicity = 1 - s1;
data = period + 0.5f * ( s0 - s2 ) / ( s0 - 2.0f * s1 + s2 );
return 0;
}
}
return _tau + 1;
}
/**
* Initialise
*
* @param threshold Allowed uncertainty
*/
void AudioAnalyzeNoteFrequency::begin( float threshold ) {
__disable_irq( );
process_buffer = false;
yin_threshold = threshold;
periodicity = 0.0f;
next_buffer = true;
running_sum = 0;
tau_global = 1;
first_run = true;
yin_idx = 1;
enabled = true;
state = 0;
data = 0.0f;
__enable_irq( );
}
/**
* available
*
* @return true if data is ready else false
*/
bool AudioAnalyzeNoteFrequency::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 AudioAnalyzeNoteFrequency::read( void ) {
__disable_irq( );
float d = data;
__enable_irq( );
return AUDIO_SAMPLE_RATE_EXACT / d;
}
/**
* Periodicity of the sampled signal from Yin algorithm from read function.
*
* @return periodicity
*/
float AudioAnalyzeNoteFrequency::probability( void ) {
__disable_irq( );
float p = periodicity;
__enable_irq( );
return p;
}
/**
* Initialise parameters.
*
* @param thresh Allowed uncertainty
*/
void AudioAnalyzeNoteFrequency::threshold( float p ) {
__disable_irq( );
yin_threshold = p;
__enable_irq( );
}
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