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Removed debug code, updated comments

dds
duff2013 8 yıl önce
ebeveyn
işleme
b180e5e675
1 değiştirilmiş dosya ile 29 ekleme ve 47 silme
  1. +29
    -47
      analyze_notefreq.cpp

+ 29
- 47
analyze_notefreq.cpp Dosyayı Görüntüle

@@ -26,22 +26,20 @@

#define HALF_BLOCKS AUDIO_GUITARTUNER_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)

/**
* 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;
@@ -54,7 +52,6 @@ void AudioAnalyzeNoteFrequency::update( void ) {
return;
}
digitalWriteFast(2, HIGH);
if ( next_buffer ) {
blocklist1[state++] = block;
if ( !first_run && process_buffer ) process( );
@@ -76,12 +73,17 @@ void AudioAnalyzeNoteFrequency::update( void ) {
process_buffer = true;
first_run = false;
state = 0;
//digitalWriteFast(LED_BUILTIN, !digitalReadFast(LED_BUILTIN));
}
}

/**
* 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.
*/
FASTRUN void AudioAnalyzeNoteFrequency::process( void ) {
//digitalWriteFast(0, HIGH);
const int16_t *p;
p = AudioBuffer;
@@ -91,52 +93,35 @@ FASTRUN void AudioAnalyzeNoteFrequency::process( void ) {
do {
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;
lag = *( ( int16_t * )p + ( x+tau ) );
current = *( ( int16_t * )p+x );
delta = ( current-lag );
sum += delta * delta;
#if F_CPU == 144000000
x += 8;
x += 8;
#elif F_CPU == 120000000
x += 12;
x += 12;
#elif F_CPU == 96000000
x += 16;
x += 16;
#elif F_CPU < 96000000
x += 32;
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 );
@@ -147,20 +132,18 @@ FASTRUN void AudioAnalyzeNoteFrequency::process( void ) {
yin_idx = 1;
running_sum = 0;
tau_global = 1;
//digitalWriteFast(0, LOW);
return;
}
tau_global = tau;
//digitalWriteFast(0, LOW);
}

/**
* check the sampled data for fundmental frequency
* 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
* @param tau lag we are currently working on gets incremented
*
* @return tau
*/
@@ -200,7 +183,6 @@ uint16_t AudioAnalyzeNoteFrequency::estimate( int64_t *yin, int64_t *rs, uint16_
* Initialise
*
* @param threshold Allowed uncertainty
* @param cpu_max How much cpu usage before throttling
*/
void AudioAnalyzeNoteFrequency::begin( float threshold ) {
__disable_irq( );

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