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update- see revision.md

main
duff2013 10 年之前
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共有 7 個文件被更改，包括 176 次插入 和 228 次删除
分割檢視 顯示文件統計
  1. +117
    -146
      AudioTuner.cpp
  2. +37
    -39
      AudioTuner.h
  3. +13
    -30
      README.md
  4. +2
    -6
      examples/Sample_Guitar_Tunning_Notes/Sample_Guitar_Tunning_Notes.ino
  5. +2
    -6
      examples/Simple_Sine/Simple_Sine.ino
  6. +1
    -1
      library.properties
  7. +4
    -0
      revision.md

+ 117
- 146
AudioTuner.cpp 查看文件

@@ -22,164 +22,136 @@

#include "AudioTuner.h"
#include "utility/dspinst.h"
#include "arm_math.h"

#if SAMPLE_RATE == SAMPLE_RATE_44100
#define SAMPLE_RATE_EXACT AUDIO_SAMPLE_RATE_EXACT / 1
#elif SAMPLE_RATE == SAMPLE_RATE_22050
#define SAMPLE_RATE_EXACT AUDIO_SAMPLE_RATE_EXACT / 2
#elif SAMPLE_RATE == SAMPLE_RATE_11025
#define SAMPLE_RATE_EXACT AUDIO_SAMPLE_RATE_EXACT / 4
#endif

#define HALF_BUFFER NUM_SAMPLES / 2
#define HALF_BLOCKS AUDIO_BLOCKS * 64

#define LOOP1(a) a
#define LOOP2(a) a LOOP1(a)
#define LOOP3(a) a LOOP2(a)
#define LOOP4(a) a LOOP3(a)
#define LOOP8(a) a LOOP3(a) a LOOP3(a)
#define LOOP16(a) a LOOP8(a) a LOOP2(a) a LOOP3(a)
#define LOOP32(a) a LOOP16(a) a LOOP8(a) a LOOP1(a) a LOOP3(a)
#define LOOP64(a) a LOOP32(a) a LOOP16(a) a LOOP8(a) a LOOP2(a) a LOOP1(a)
#define UNROLL(n,a) LOOP##n(a)

/**
* Audio update function.
*/
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++;
}

void AudioTuner::update( void ) {
audio_block_t *block;
const int16_t *p, *end;
block = receiveReadOnly( );
if ( !block ) return;
block = receiveReadOnly();
if (!block) return;
if ( !enabled ) {
release( block );
return;
}
p = block->data;
end = p + AUDIO_BLOCK_SAMPLES;
/*
* Double buffering, one fills while the other is processed
* 2x the throughput.
*/
uint16_t *dst;
bool next = next_buffer;
if ( next ) {
//digitalWriteFast(6, HIGH);
dst = ( uint16_t * )buffer;
digitalWriteFast(2, HIGH);
if ( next_buffer ) {
blocklist1[state++] = block;
if ( !first_run && process_buffer ) process( );
} else {
blocklist2[state++] = block;
if ( !first_run && process_buffer ) process( );
}
else {
//digitalWriteFast(6, LOW);
dst = ( uint16_t * )buffer + NUM_SAMPLES;
if ( state >= AUDIO_BLOCKS ) {
if ( next_buffer ) {
if ( !first_run && process_buffer ) process( );
for ( int i = 0; i < AUDIO_BLOCKS; i++ ) copy_buffer( AudioBuffer+( i * 0x80 ), blocklist1[i]->data );
for ( int i = 0; i < AUDIO_BLOCKS; i++ ) release(blocklist1[i] );
} else {
if ( !first_run && process_buffer ) process( );
for ( int i = 0; i < AUDIO_BLOCKS; i++ ) copy_buffer( AudioBuffer+( i * 0x80 ), blocklist2[i]->data );
for ( int i = 0; i < AUDIO_BLOCKS; i++ ) release( blocklist2[i] );
}
process_buffer = true;
first_run = false;
state = 0;
//digitalWriteFast(LED_BUILTIN, !digitalReadFast(LED_BUILTIN));
}
}

FASTRUN void AudioTuner::process( void ) {
//digitalWriteFast(0, HIGH);
// gather data/and release block
uint16_t count = count_global;
const int16_t *p;
p = AudioBuffer;
uint16_t cycles = 64;;
uint16_t tau = tau_global;
do {
*( dst+count++ ) = *( uint16_t * )p;
p += SAMPLE_RATE;
} while ( p < end );
release( block );
uint16_t x = 0;
int64_t sum = 0;
//uint32_t res;
do {
/*int16_t current1, lag1, current2, lag2;
int32_t val1, val2;
lag1 = *( ( uint32_t * )p + ( x + tau ) );
current1 = *( ( uint32_t * )p + x );
x += 32;
lag2 = *( ( uint32_t * )p + ( x + tau ) );
current2 = *( ( uint32_t * )p + x );
val1 = __PKHBT(current1, current2, 0x10);
val2 = __PKHBT(lag1, lag2, 0x10);
res = __SSUB16( val1, val2 );
sum = __SMLALD(res, res, sum);
//sum = __SMLSLD(delta1, delta2, sum);*/
int16_t current, lag, delta;
//UNROLL(16,
lag = *( ( int16_t * )p + ( x+tau ) );
current = *( ( int16_t * )p+x );
delta = ( current-lag );
sum += delta * delta;
#if F_CPU == 144000000
x += 8;
#elif F_CPU == 120000000
x += 12;
#elif F_CPU == 96000000
x += 16;
#elif F_CPU < 96000000
x += 32;
#endif
//);
} while ( x <= HALF_BLOCKS );

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;
yin_idx = 1;
running_sum = 0;
tau_global = 1;
//digitalWriteFast(2, LOW);
//digitalWriteFast(0, LOW);
return;
}
} while ( --cycles );
/*
* If buffer full switch to start filling next
* buffer and process the just filled buffer.
*/
if ( count >= NUM_SAMPLES ) {
//digitalWriteFast(2, !digitalReadFast(2));
__disable_irq();
next_buffer = !next_buffer;
process_buffer = true;
count_global = 0;
tau_global = 1;
if ( tau >= HALF_BLOCKS ) {
process_buffer = false;
new_output = false;
yin_idx = 1;
running_sum = 0;
count = 0;
__enable_irq();
}
count_global = count;// update global count
/*
* Set the number of cycles to be processed per receiving block.
*/
uint16_t cycles;
const uint16_t usage_max = cpu_usage_max;
if ( AudioProcessorUsage( ) > usage_max ) {
#if NUM_SAMPLES >= 8192
cycles = tau_global + 2;
#elif NUM_SAMPLES == 4096
cycles = tau_global + 4;
#elif NUM_SAMPLES == 2048
cycles = tau_global + 8;
#elif NUM_SAMPLES <= 1024
cycles = tau_global + 32;
#endif
}
else {
#if NUM_SAMPLES >= 8192
cycles = tau_global + 8;
#elif NUM_SAMPLES == 4096
cycles = tau_global + 16;
#elif NUM_SAMPLES == 2048
cycles = tau_global + 32;
#elif NUM_SAMPLES <= 1024
cycles = tau_global + 64;
#endif
}
if ( process_buffer ) {
//digitalWriteFast(0, HIGH);
uint16_t tau;
next = next_buffer;
tau = tau_global;
do {
int64_t sum = 0;
const int16_t *end, *buf;
if ( next ) {
//digitalWriteFast(4, LOW);
buf = buffer + NUM_SAMPLES;
}
else {
//digitalWriteFast(4, HIGH);
buf = buffer;
}
end = buf + HALF_BUFFER;
// TODO: How to make faster?
do {
int16_t current, lag, delta;
UNROLL( 8,
lag = *( buf + tau );
current = *buf++;
delta = current - lag;
//sum = multiply_accumulate_32x32_rshift32_rounded(sum, delta, delta);
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 <= cycles );
tau_global = tau;
tau_global = 1;
//digitalWriteFast(0, LOW);
return;
}
tau_global = tau;
//digitalWriteFast(0, LOW);
}

/**
@@ -193,9 +165,10 @@ void AudioTuner::update( void ) {
* @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 *y = ( int64_t * )yin;
const int64_t *r = ( int64_t * )rs;
uint16_t _tau, _head;
const float thresh = yin_threshold;
_tau = tau;
_head = head;
@@ -209,19 +182,16 @@ uint16_t AudioTuner::estimate( int64_t *yin, int64_t *rs, uint16_t head, uint16_
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 ) );
s0 = ( ( float )*( y+idx0 ) / *( r+idx0 ) );
s1 = ( ( float )*( y+idx1 ) / *( r+idx1 ) );
s2 = ( ( float )*( y+idx2 ) / *( r+idx2 ) );
if ( s1 < yin_threshold && s1 < s2 ) {
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;
}
//if ( s1 > 2.4 ) return _tau + 2;
//else return _tau + 1;
}
return _tau + 1;
}
@@ -232,18 +202,19 @@ uint16_t AudioTuner::estimate( int64_t *yin, int64_t *rs, uint16_t head, uint16_
* @param threshold Allowed uncertainty
* @param cpu_max How much cpu usage before throttling
*/
void AudioTuner::initialize( float threshold, float cpu_max ) {
void AudioTuner::initialize( float threshold ) {
__disable_irq( );
cpu_usage_max = cpu_max*100;
yin_threshold = threshold;
process_buffer = false;
yin_threshold = threshold;
periodicity = 0.0f;
next_buffer = true;
running_sum = 0;
count_global = 0;
tau_global = 1;
first_run = true;
yin_idx = 1;
data = 0;
enabled = true;
state = 0;
data = 0.0f;
__enable_irq( );
}

@@ -269,7 +240,7 @@ float AudioTuner::read( void ) {
__disable_irq( );
float d = data;
__enable_irq( );
return SAMPLE_RATE_EXACT / d;
return AUDIO_SAMPLE_RATE_EXACT / d;
}

/**

+ 37
- 39
AudioTuner.h 查看文件

@@ -24,62 +24,46 @@
#define AudioTuner_h_

#include "AudioStream.h"
/****************************************************************/
#define SAMPLE_RATE_44100 1 // 44100 sample rate
#define SAMPLE_RATE_22050 2 // 22050 sample rate
#define SAMPLE_RATE_11025 4 // 11025 sample rate
/****************************************************************/

/****************************************************************
* Safe to adjust these values below *
* *
* These two parameters define how this object works. *
* This parameter defines the size of the buffer. *
* *
* 1. NUM_SAMPLES - Size of the buffer. Since object uses *
* double buffering this value will be 4x in bytes of *
* memory. !!! Must be power of 2 !!!! *
* 1. AUDIO_BLOCKS - Buffer size is 128 * AUDIO_BLOCKS. *
* The more AUDIO_BLOCKS the lower the *
* frequency you can detect. The defualt *
* (24) is set to measure down to 29.14 *
* Hz or B(flat)0. *
* *
* 2. SAMPLE_RATE - Just what it says. *
* *
* These two parameters work hand in hand. For example if you *
* want a high sample rate but do not allocate enough buffer *
* space, you will be limit how low of a frequency you can *
* measure. If you then increase the buffer you use up *
* precious ram and slow down the system since it takes longer *
* to processes the buffer. *
* *
* Play around with these values to find what best suits your *
* needs. The max number of buffers you can have is 8192 bins. *
****************************************************************/
// !!! Must be power of 2 !!!!
#define NUM_SAMPLES 2048 // make a power of two

// Use defined sample rates above^
#define SAMPLE_RATE SAMPLE_RATE_22050
#define AUDIO_BLOCKS 24
/****************************************************************/

class AudioTuner : public AudioStream
{
class AudioTuner : public AudioStream {
public:
/**
* constructor to setup Audio Library and initialize
*
* @return none
*/
AudioTuner( void ) : AudioStream( 1, inputQueueArray ), enabled( false ), new_output(false) {}
AudioTuner( void ) : AudioStream( 1, inputQueueArray ), enabled( false ), new_output(false) {
}
/**
* initialize variables and start conversion
*
* @param threshold Allowed uncertainty
* @param cpu_max How much cpu usage before throttling
*
* @return none
*/
void initialize( float threshold, float cpu_max);
void initialize( float threshold );
/**
* sets threshold value
*
* @param thresh
* @return none
*/
void threshold( float p );
@@ -105,9 +89,11 @@ public:
/**
* Audio Library calls this update function ~2.9ms
*
* @return none
*/
virtual void update( void );
private:
/**
* check the sampled data for fundamental frequency
@@ -121,14 +107,26 @@ private:
*/
uint16_t estimate( int64_t *yin, int64_t *rs, uint16_t head, uint16_t tau );
int16_t buffer[NUM_SAMPLES*2] __attribute__ ( ( aligned ( 4 ) ) );
float periodicity, yin_threshold, data, cpu_usage_max;
int64_t rs_buffer[5], yin_buffer[5];
/**
* process audio data
*
* @return none
*/
void process( void );
/**
* Variables
*/
uint64_t running_sum;
uint16_t tau_global, count_global, tau_cycles;
uint8_t yin_idx;
bool enabled, process_buffer, next_buffer;
volatile bool new_output;
uint16_t tau_global;
int64_t rs_buffer[5], yin_buffer[5];
int16_t AudioBuffer[AUDIO_BLOCKS*128] __attribute__ ( ( aligned ( 4 ) ) );
uint8_t yin_idx, state;
float periodicity, yin_threshold, cpu_usage_max, data;
bool enabled, next_buffer, first_run;
volatile bool new_output, process_buffer;
audio_block_t *blocklist1[AUDIO_BLOCKS];
audio_block_t *blocklist2[AUDIO_BLOCKS];
audio_block_t *inputQueueArray[1];
};
#endif

+ 13
- 30
README.md 查看文件

@@ -1,5 +1,5 @@
<p align="center">
<b>Guitar and Bass Tuner Library v2.2</b><br>
<b>Guitar and Bass Tuner Library v2.3</b><br>
<b>Teensy 3.1/2</b><br>
</p>

@@ -40,46 +40,28 @@
*---<\ P /
\_/

>Many optimizations have been done to the [YIN] algorithm for frequencies between 29-360Hz.
>Many optimizations have been done to the [YIN] algorithm for frequencies between 29-400Hz.
>>While its still using a brute force method ( n<sup>2</sup> ) for finding the fundamental frequency f<sub>o</sub>, it is tuned to skip certain <b>tau</b> (<img src="http://latex.numberempire.com/render?%5Cinline%20%5Chuge%20%5Cmathbf%7B%5Ctau%7D&sig=845639da85c0dd8e2de679817b06639c"/></img>) values and focus mostly on frequencies found in the bass and guitar.
>>>The input is double buffered so while you are processing one buffer it is filling the other to double throughput.
>>>>There are a few parameters that can be adjusted to "dial in" the algorithm for better estimations located in AudioTuner.h. The defaults below are what I found that have the best trade off for speed and accuracy.
>>>>The parameter AUDIO_BLOCKS below can be adjusted but its default of 24 I found to be best to work with the guitar and bass frequency range (29- 400)Hz.
>>>>Looking into finding the Auto Correlation using FFT and IFFT to speed up processing of data! Not that simple because the YIN algorithm uses a squared difference tweak to the Auto Correlation.

<h4>AudioTuner.h</h4>

```
/****************************************************************/
#define SAMPLE_RATE_44100 1 // 44100 sample rate
#define SAMPLE_RATE_22050 2 // 22050 sample rate
#define SAMPLE_RATE_11025 4 // 11025 sample rate
/****************************************************************/

/****************************************************************
* Safe to adjust these values below *
* *
* These two parameters define how this object works. *
* This parameter defines the size of the buffer. *
* *
* 1. NUM_SAMPLES - Size of the buffer. Since object uses *
* double buffering this value will be 4x in bytes of *
* memory. !!! Must be power of 2 !!!! *
* 1. AUDIO_BLOCKS - Buffer size is 128 * AUDIO_BLOCKS. *
* The more AUDIO_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. SAMPLE_RATE - Just what it says. *
* *
* These two parameters work hand in hand. For example if you *
* want a high sample rate but do not allocate enough buffer *
* space, you will be limit how low of a frequency you can *
* measure. If you then increase the buffer you use up *
* precious ram and slow down the system since it takes longer *
* to processes the buffer. *
* *
* Play around with these values to find what best suits your *
* needs. The max number of buffers you can have is 8192 bins. *
****************************************************************/
// !!! Must be power of 2 !!!!
#define NUM_SAMPLES 2048 // make a power of two

// Use defined sample rates above^
#define SAMPLE_RATE SAMPLE_RATE_22050
#define AUDIO_BLOCKS 24
/****************************************************************/
```

@@ -94,4 +76,5 @@
</ol>
</div>

[YIN]:http://recherche.ircam.fr/equipes/pcm/cheveign/pss/2002_JASA_YIN.pdf
[YIN]:http://recherche.ircam.fr/equipes/pcm/cheveign/pss/2002_JASA_YIN.pdf
[Teensy Audio Library]:http://www.pjrc.com/teensy/td_libs_Audio.html

+ 2
- 6
examples/Sample_Guitar_Tunning_Notes/Sample_Guitar_Tunning_Notes.ino 查看文件

@@ -57,16 +57,12 @@ void playNote(void) {
}
//---------------------------------------------------------------------------------------
void setup() {
AudioMemory(4);
AudioMemory(30);
/*
* Initialize the yin algorithm's absolute
* threshold, this is good number.
*
* Percent of overall current cpu usage used
* before making the search algorithm less
* aggressive (0.0 - 1.0).
*/
tuner.initialize(.15, .99);
tuner.initialize(.15);
pinMode(LED_BUILTIN, OUTPUT);
playNoteTimer.begin(playNote, 1000);
}

+ 2
- 6
examples/Simple_Sine/Simple_Sine.ino 查看文件

@@ -40,16 +40,12 @@ AudioConnection patchCord3(mixer, 0, dac, 0);
char buffer[10];

void setup() {
AudioMemory(4);
AudioMemory(30);
/*
* Initialize the yin algorithm's absolute
* threshold, this is good number.
*
* Percent of overall current cpu usage used
* before making the search algorithm less
* aggressive (0.0 - 1.0).
*/
tuner.initialize(.15, .99);
tuner.initialize(.15);
sine.frequency(30.87);
sine.amplitude(1);

+ 1
- 1
library.properties 查看文件

@@ -1,5 +1,5 @@
name=AudioTuner
version=2.2
version=2.3
author=Colin Duffy
maintainer=Colin Duffy
sentence=Yin algorithm

+ 4
- 0
revision.md 查看文件

@@ -1,3 +1,7 @@
><b>Updated (11/23/15 v2.3)</b><br>
* Totally new method to gather and process data, data is available after 24 Blocks of data have been collected (~69.6ms) for all frequencies.<br>
* Double buffer to collect Audio data, while one collects the other buffer is processed.<br>

><b>Updated (10/12/15 v2.2)</b><br>
* Fixed yin cpu usage throttling code in update function.<br>
* Function initialize second param takes a float (0.0 - 1.0).<br>

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