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rockbox/apps/plugins/pitch_detector.c
Teruaki Kawashima 01313d5039 * Rename backlight_force_on to backlight_ignore_timeout to make it clear what the function does. 
* Add backlight_force_on() which forces to turn on backlight even when the setting is set to Off. use this in lamp.
No functional change except lamp.
See aslo FS#9883.


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29128 a1c6a512-1295-4272-9138-f99709370657
2011-01-24 12:29:16 +00:00

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/**************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2008 Lechner Michael / smoking gnu
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
* ----------------------------------------------------------------------------
*
* INTRODUCTION:
* OK, this is an attempt to write an instrument tuner for rockbox.
* It uses a Schmitt trigger algorithm, which I copied from
* tuneit [ (c) 2004 Mario Lang <mlang@delysid.org> ], for detecting the
* fundamental freqency of a sound. A FFT algorithm would be more accurate
* but also much slower.
*
* TODO:
* - Adapt the Yin FFT algorithm, which would reduce complexity from O(n^2)
* to O(nlogn), theoretically reducing latency by a factor of ~10. -David
*
* MAJOR CHANGES:
* 08.03.2008 Started coding
* 21.03.2008 Pitch detection works more or less
* Button definitions for most targets added
* 02.04.2008 Proper GUI added
* Todo, Major Changes and Current Limitations added
* 08.19.2009 Brought the code up to date with current plugin standards
* Made it work more nicely with color, BW and grayscale
* Changed pitch detection to use the Yin algorithm (better
* detection, but slower -- would be ~4x faster with
* fixed point math, I think). Code was poached from the
* Aubio sound processing library (aubio.org). -David
* 08.31.2009 Lots of changes:
* Added a menu to tweak settings
* Converted everything to fixed point (greatly improving
* latency)
* Improved the display
* Improved efficiency with judicious use of cpu_boost, the
* backlight, and volume detection to limit unneeded
* calculation
* Fixed a problem that caused an octave-off error
* -David
* 05.14.2010 Multibuffer continuous recording with two buffers
*
*
* CURRENT LIMITATIONS:
* - No gapless recording. Strictly speaking true gappless isn't possible,
* since the algorithm takes longer to calculate than the length of the
* sample, but latency could be improved a bit with proper use of the DMA
* recording functions.
* - Due to how the Yin algorithm works, latency is higher for lower
* frequencies.
*/
#include "plugin.h"
#include "lib/pluginlib_actions.h"
#include "lib/picture.h"
#include "lib/helper.h"
#include "pluginbitmaps/pitch_notes.h"
/* Some fixed point calculation stuff */
typedef int32_t fixed;
#define FIXED_PRECISION 18
#define FP_MAX ((fixed) {0x7fffffff})
#define FP_MIN ((fixed) {-0x80000000})
#define int2fixed(x) ((fixed)((x) << FIXED_PRECISION))
#define int2mantissa(x) ((fixed)(x))
#define fixed2int(x) ((int)((x) >> FIXED_PRECISION))
#define fixed2float(x) (((float)(x)) / ((float)(1 << FIXED_PRECISION)))
#define float2fixed(x) ((fixed)(x * (float)(1 << FIXED_PRECISION)))
/* I adapted these ones from the Rockbox fixed point library */
#define fp_mul(x, y) \
((fixed)((((int64_t)((x))) * ((int64_t)((y)))) >> (FIXED_PRECISION)))
#define fp_div(x, y) \
((fixed)((((int64_t)((x))) << (FIXED_PRECISION)) / ((int64_t)((y)))))
/* Operators for fixed point */
#define fp_add(x, y) ((fixed)((x) + (y)))
#define fp_sub(x, y) ((fixed)((x) - (y)))
#define fp_shl(x, y) ((fixed)((x) << (y)))
#define fp_shr(x, y) ((fixed)((x) >> (y)))
#define fp_neg(x) ((fixed)(-(x)))
#define fp_gt(x, y) ((x) > (y))
#define fp_gte(x, y) ((x) >= (y))
#define fp_lt(x, y) ((x) < (y))
#define fp_lte(x, y) ((x) <= (y))
#define fp_sqr(x) fp_mul((x), (x))
#define fp_equal(x, y) ((x) == (y))
#define fp_round(x) (fixed2int(fp_add((x), float2fixed(0.5))))
#define fp_data(x) (x)
#define fp_frac(x) (fp_sub((x), int2fixed(fixed2int(x))))
#define FP_ZERO ((fixed)0)
#define FP_LOW ((fixed)2)
/* Some defines for converting between period and frequency */
/* I introduce some divisors in this because the fixed point */
/* variables aren't big enough to hold higher than a certain */
/* value. This loses a bit of precision but it means we */
/* don't have to use 32.32 variables (yikes). */
/* With an 18-bit decimal precision, the max value in the */
/* integer part is 8192. Divide 44100 by 7 and it'll fit in */
/* that variable. */
#define fp_period2freq(x) fp_div(int2fixed(sample_rate / 7), \
fp_div((x),int2fixed(7)))
#define fp_freq2period(x) fp_period2freq(x)
#define period2freq(x) (sample_rate / (x))
#define freq2period(x) period2freq(x)
#define sqr(x) ((x)*(x))
/* Some constants for tuning */
#define A_FREQ float2fixed(440.0f)
#define D_NOTE float2fixed(1.059463094359f)
#define LOG_D_NOTE float2fixed(1.0f/12.0f)
#define D_NOTE_SQRT float2fixed(1.029302236643f)
#define LOG_2 float2fixed(1.0f)
/* The recording buffer size */
/* This is how much is sampled at a time. */
/* It also determines latency -- if BUFFER_SIZE == sample_rate then */
/* there'll be one sample per second, or a latency of one second. */
/* Furthermore, the lowest detectable frequency will be about twice */
/* the number of reads per second */
/* If we ever switch to Yin FFT algorithm then this needs to be
a power of 2 */
#define BUFFER_SIZE 4096
#define SAMPLE_SIZE 4096
#define SAMPLE_SIZE_MIN 1024
#define YIN_BUFFER_SIZE (BUFFER_SIZE / 4)
#define LCD_FACTOR (fp_div(int2fixed(LCD_WIDTH), int2fixed(100)))
/* The threshold for the YIN algorithm */
#define DEFAULT_YIN_THRESHOLD 5 /* 0.10 */
static const fixed yin_threshold_table[] IDATA_ATTR =
{
float2fixed(0.01),
float2fixed(0.02),
float2fixed(0.03),
float2fixed(0.04),
float2fixed(0.05),
float2fixed(0.10),
float2fixed(0.15),
float2fixed(0.20),
float2fixed(0.25),
float2fixed(0.30),
float2fixed(0.35),
float2fixed(0.40),
float2fixed(0.45),
float2fixed(0.50),
};
/* Structure for the reference frequency (frequency of A)
* It's used for scaling the frequency before finding out
* the note. The frequency is scaled in a way that the main
* algorithm can assume the frequency of A to be 440 Hz.
*/
static const struct
{
const int frequency; /* Frequency in Hz */
const fixed ratio; /* 440/frequency */
const fixed logratio; /* log2(factor) */
} freq_A[] =
{
{435, float2fixed(1.011363636), float2fixed( 0.016301812)},
{436, float2fixed(1.009090909), float2fixed( 0.013056153)},
{437, float2fixed(1.006818182), float2fixed( 0.009803175)},
{438, float2fixed(1.004545455), float2fixed( 0.006542846)},
{439, float2fixed(1.002272727), float2fixed( 0.003275132)},
{440, float2fixed(1.000000000), float2fixed( 0.000000000)},
{441, float2fixed(0.997727273), float2fixed(-0.003282584)},
{442, float2fixed(0.995454545), float2fixed(-0.006572654)},
{443, float2fixed(0.993181818), float2fixed(-0.009870244)},
{444, float2fixed(0.990909091), float2fixed(-0.013175389)},
{445, float2fixed(0.988636364), float2fixed(-0.016488123)},
};
/* Index of the entry for 440 Hz in the table (default frequency for A) */
#define DEFAULT_FREQ_A 5
#define NUM_FREQ_A (sizeof(freq_A)/sizeof(freq_A[0]))
/* How loud the audio has to be to start displaying pitch */
/* Must be between 0 and 100 */
#define VOLUME_THRESHOLD (50)
/* Change to AUDIO_SRC_LINEIN if you want to record from line-in */
#ifdef HAVE_MIC_IN
#define INPUT_TYPE AUDIO_SRC_MIC
#else
#define INPUT_TYPE AUDIO_SRC_LINEIN
#endif
/* How many decimal places to display for the Hz value */
#define DISPLAY_HZ_PRECISION 100
/* Where to put the various GUI elements */
static int note_y;
static int bar_grad_y;
#define LCD_RES_MIN (LCD_HEIGHT < LCD_WIDTH ? LCD_HEIGHT : LCD_WIDTH)
#define BAR_PADDING (LCD_RES_MIN / 32)
#define BAR_Y (LCD_HEIGHT * 3 / 4)
#define BAR_HEIGHT (LCD_RES_MIN / 4 - BAR_PADDING)
#define BAR_HLINE_Y (BAR_Y - BAR_PADDING)
#define BAR_HLINE_Y2 (BAR_Y + BAR_HEIGHT + BAR_PADDING - 1)
#define HZ_Y 0
#define GRADUATION 10 /* Subdivisions of the whole 100-cent scale */
/* Bitmaps for drawing the note names. These need to have height
<= (bar_grad_y - note_y), or 15/32 * LCD_HEIGHT
*/
#define NUM_NOTE_IMAGES 9
#define NOTE_INDEX_A 0
#define NOTE_INDEX_B 1
#define NOTE_INDEX_C 2
#define NOTE_INDEX_D 3
#define NOTE_INDEX_E 4
#define NOTE_INDEX_F 5
#define NOTE_INDEX_G 6
#define NOTE_INDEX_SHARP 7
#define NOTE_INDEX_FLAT 8
static const struct picture note_bitmaps =
{
pitch_notes,
BMPWIDTH_pitch_notes,
BMPHEIGHT_pitch_notes,
BMPHEIGHT_pitch_notes/NUM_NOTE_IMAGES
};
static unsigned int sample_rate;
static int audio_head = 0; /* which of the two buffers to use? */
static volatile int audio_tail = 0; /* which of the two buffers to record? */
/* It's stereo, so make the buffer twice as big */
#ifndef SIMULATOR
static int16_t audio_data[2][BUFFER_SIZE] MEM_ALIGN_ATTR;
static fixed yin_buffer[YIN_BUFFER_SIZE] IBSS_ATTR;
#ifdef PLUGIN_USE_IRAM
static int16_t iram_audio_data[BUFFER_SIZE] IBSS_ATTR;
#else
#define iram_audio_data audio_data[audio_head]
#endif
#endif
/* Notes within one (reference) scale */
static const struct
{
const char *name; /* Name of the note, e.g. "A#" */
const fixed freq; /* Note frequency, Hz */
const fixed logfreq; /* log2(frequency) */
} notes[] =
{
{"A" , float2fixed(440.0000000f), float2fixed(8.781359714f)},
{"A#", float2fixed(466.1637615f), float2fixed(8.864693047f)},
{"B" , float2fixed(493.8833013f), float2fixed(8.948026380f)},
{"C" , float2fixed(523.2511306f), float2fixed(9.031359714f)},
{"C#", float2fixed(554.3652620f), float2fixed(9.114693047f)},
{"D" , float2fixed(587.3295358f), float2fixed(9.198026380f)},
{"D#", float2fixed(622.2539674f), float2fixed(9.281359714f)},
{"E" , float2fixed(659.2551138f), float2fixed(9.364693047f)},
{"F" , float2fixed(698.4564629f), float2fixed(9.448026380f)},
{"F#", float2fixed(739.9888454f), float2fixed(9.531359714f)},
{"G" , float2fixed(783.9908720f), float2fixed(9.614693047f)},
{"G#", float2fixed(830.6093952f), float2fixed(9.698026380f)},
};
/* GUI */
#if LCD_DEPTH > 1
static unsigned front_color;
#endif
static int font_w,font_h;
static int bar_x_0;
static int lbl_x_minus_50, lbl_x_minus_20, lbl_x_0, lbl_x_20, lbl_x_50;
/* Settings for the plugin */
static struct tuner_settings
{
unsigned volume_threshold;
unsigned record_gain;
unsigned sample_size;
unsigned lowest_freq;
unsigned yin_threshold;
int freq_A; /* Index of the frequency of A */
bool use_sharps;
bool display_hz;
int key_transposition; /* Which note to display as 'C'. */
/* 0=C, 1=D-flat, 2=D, ..., 11=B. This is useful if you */
/* use a transposing instrument. In that case, this */
/* setting tells which 'real' note is played by the */
/* instrument if you play a written 'C'. Thus, this */
/* setting is the number of semitones from the real 'C' */
/* up to the 'instrument key'. */
} settings;
/* By default, the real 'C' is displayed as 'C' */
#define DEFAULT_KEY_TRANSPOSITION 0
/*=================================================================*/
/* Settings loading and saving(adapted from the clock plugin) */
/*=================================================================*/
#define SETTINGS_FILENAME PLUGIN_APPS_DIR "/.pitch_detector_settings"
/* The settings as they exist on the hard disk, so that
* we can know at saving time if changes have been made */
static struct tuner_settings hdd_settings;
/*---------------------------------------------------------------------*/
static bool settings_needs_saving(void)
{
return(rb->memcmp(&settings, &hdd_settings, sizeof(settings)));
}
/*---------------------------------------------------------------------*/
static void tuner_settings_reset(void)
{
settings = (struct tuner_settings) {
.volume_threshold = VOLUME_THRESHOLD,
.record_gain = rb->global_settings->rec_mic_gain,
.sample_size = BUFFER_SIZE,
.lowest_freq = period2freq(BUFFER_SIZE / 4),
.yin_threshold = DEFAULT_YIN_THRESHOLD,
.freq_A = DEFAULT_FREQ_A,
.use_sharps = true,
.display_hz = false,
.key_transposition = DEFAULT_KEY_TRANSPOSITION,
};
}
/*---------------------------------------------------------------------*/
static void load_settings(void)
{
int fd = rb->open(SETTINGS_FILENAME, O_RDONLY);
if(fd < 0){ /* file doesn't exist */
/* Initializes the settings with default values at least */
tuner_settings_reset();
return;
}
/* basic consistency check */
if(rb->filesize(fd) == sizeof(settings)){
rb->read(fd, &settings, sizeof(settings));
rb->memcpy(&hdd_settings, &settings, sizeof(settings));
}
else{
tuner_settings_reset();
}
rb->close(fd);
}
/*---------------------------------------------------------------------*/
static void save_settings(void)
{
if(!settings_needs_saving())
return;
int fd = rb->creat(SETTINGS_FILENAME, 0666);
if(fd >= 0){ /* does file exist? */
rb->write (fd, &settings, sizeof(settings));
rb->close(fd);
}
}
/*=================================================================*/
/* MENU */
/*=================================================================*/
/* Keymaps */
const struct button_mapping* plugin_contexts[]={
pla_main_ctx,
#if NB_SCREENS == 2
pla_remote_ctx,
#endif
};
#define PLA_ARRAY_COUNT sizeof(plugin_contexts)/sizeof(plugin_contexts[0])
/* Option strings */
/* This has to match yin_threshold_table */
static const struct opt_items yin_threshold_text[] =
{
{ "0.01", -1 },
{ "0.02", -1 },
{ "0.03", -1 },
{ "0.04", -1 },
{ "0.05", -1 },
{ "0.10", -1 },
{ "0.15", -1 },
{ "0.20", -1 },
{ "0.25", -1 },
{ "0.30", -1 },
{ "0.35", -1 },
{ "0.40", -1 },
{ "0.45", -1 },
{ "0.50", -1 },
};
static const struct opt_items accidental_text[] =
{
{ "Flat", -1 },
{ "Sharp", -1 },
};
static const struct opt_items transpose_text[] =
{
{ "C (Concert Pitch)", -1 },
{ "D-flat", -1 },
{ "D", -1 },
{ "E-flat", -1 },
{ "E", -1 },
{ "F", -1 },
{ "G-flat", -1 },
{ "G", -1 },
{ "A-flat", -1 },
{ "A", -1 },
{ "B-flat", -1 },
{ "B", -1 },
};
static void set_min_freq(int new_freq)
{
settings.sample_size = freq2period(new_freq) * 4;
/* clamp the sample size between min and max */
if(settings.sample_size <= SAMPLE_SIZE_MIN)
settings.sample_size = SAMPLE_SIZE_MIN;
else if(settings.sample_size >= BUFFER_SIZE)
settings.sample_size = BUFFER_SIZE;
/* sample size must be divisible by 4 - round up */
settings.sample_size = (settings.sample_size + 3) & ~3;
}
/* Displays the menu. Returns true iff the user selects 'quit'. */
static bool main_menu(void)
{
int selection = 0;
bool done = false;
bool exit_tuner = false;
int choice;
int freq_val;
bool reset;
backlight_use_settings();
#ifdef HAVE_SCHEDULER_BOOSTCTRL
rb->cancel_cpu_boost();
#endif
MENUITEM_STRINGLIST(menu,"Tuner Settings",NULL,
"Return to Tuner",
"Volume Threshold",
"Listening Volume",
"Lowest Frequency",
"Algorithm Pickiness",
"Accidentals",
"Key Transposition",
"Display Frequency (Hz)",
"Frequency of A (Hz)",
"Reset Settings",
"Quit");
while(!done)
{
choice = rb->do_menu(&menu, &selection, NULL, false);
switch(choice)
{
case 1:
rb->set_int("Volume Threshold", "%", UNIT_INT,
&settings.volume_threshold,
NULL, 5, 5, 95, NULL);
break;
case 2:
rb->set_int("Listening Volume", "%", UNIT_INT,
&settings.record_gain,
NULL, 1, rb->sound_min(SOUND_MIC_GAIN),
rb->sound_max(SOUND_MIC_GAIN), NULL);
break;
case 3:
rb->set_int("Lowest Frequency", "Hz", UNIT_INT,
&settings.lowest_freq, set_min_freq, 1,
/* Range depends on the size of the buffer */
sample_rate / (BUFFER_SIZE / 4),
sample_rate / (SAMPLE_SIZE_MIN / 4), NULL);
break;
case 4:
rb->set_option(
"Algorithm Pickiness (Lower -> more discriminating)",
&settings.yin_threshold,
INT, yin_threshold_text,
sizeof(yin_threshold_text) / sizeof(yin_threshold_text[0]),
NULL);
break;
case 5:
rb->set_option("Display Accidentals As",
&settings.use_sharps,
BOOL, accidental_text, 2, NULL);
break;
case 6:
rb->set_option("Key Transposition",
&settings.key_transposition,
INT, transpose_text, 12, NULL);
break;
case 7:
rb->set_bool("Display Frequency (Hz)",
&settings.display_hz);
break;
case 8:
freq_val = freq_A[settings.freq_A].frequency;
rb->set_int("Frequency of A (Hz)",
"Hz", UNIT_INT, &freq_val, NULL,
1, freq_A[0].frequency, freq_A[NUM_FREQ_A-1].frequency,
NULL);
settings.freq_A = freq_val - freq_A[0].frequency;
break;
case 9:
reset = false;
rb->set_bool("Reset Tuner Settings?", &reset);
if (reset)
tuner_settings_reset();
break;
case 10:
exit_tuner = true;
done = true;
break;
case 0:
default:
/* Return to the tuner */
done = true;
break;
}
}
backlight_ignore_timeout();
return exit_tuner;
}
/*=================================================================*/
/* Binary Log */
/*=================================================================*/
/* Fixed-point log base 2*/
/* Adapted from python code at
http://en.wikipedia.org/wiki/Binary_logarithm#Algorithm
*/
static fixed log(fixed inp)
{
fixed x = inp;
fixed fp = int2fixed(1);
fixed res = int2fixed(0);
if(fp_lte(x, FP_ZERO))
{
return FP_MIN;
}
/* Integer part*/
/* while x<1 */
while(fp_lt(x, int2fixed(1)))
{
res = fp_sub(res, int2fixed(1));
x = fp_shl(x, 1);
}
/* while x>=2 */
while(fp_gte(x, int2fixed(2)))
{
res = fp_add(res, int2fixed(1));
x = fp_shr(x, 1);
}
/* Fractional part */
/* while fp > 0 */
while(fp_gt(fp, FP_ZERO))
{
fp = fp_shr(fp, 1);
x = fp_mul(x, x);
/* if x >= 2 */
if(fp_gte(x, int2fixed(2)))
{
x = fp_shr(x, 1);
res = fp_add(res, fp);
}
}
return res;
}
/*=================================================================*/
/* GUI Stuff */
/*=================================================================*/
/* Draw the note bitmap */
static void draw_note(const char *note)
{
int i;
int note_x = (LCD_WIDTH - BMPWIDTH_pitch_notes) / 2;
int accidental_index = NOTE_INDEX_SHARP;
i = note[0]-'A';
if(note[1] == '#')
{
if(!(settings.use_sharps))
{
i = (i + 1) % 7;
accidental_index = NOTE_INDEX_FLAT;
}
vertical_picture_draw_sprite(rb->screens[0],
&note_bitmaps,
accidental_index,
LCD_WIDTH / 2,
note_y);
note_x = LCD_WIDTH / 2 - BMPWIDTH_pitch_notes;
}
vertical_picture_draw_sprite(rb->screens[0], &note_bitmaps, i,
note_x,
note_y);
}
/* Draw the red bar and the white lines */
static void draw_bar(fixed wrong_by_cents)
{
unsigned n;
int x;
#ifdef HAVE_LCD_COLOR
rb->lcd_set_foreground(LCD_RGBPACK(255,255,255)); /* Color screens */
#elif LCD_DEPTH > 1
rb->lcd_set_foreground(LCD_BLACK); /* Greyscale screens */
#endif
rb->lcd_hline(0,LCD_WIDTH-1, BAR_HLINE_Y);
rb->lcd_hline(0,LCD_WIDTH-1, BAR_HLINE_Y2);
/* Draw graduation lines on the off-by readout */
for(n = 0; n <= GRADUATION; n++)
{
x = (LCD_WIDTH * n + GRADUATION / 2) / GRADUATION;
if (x >= LCD_WIDTH)
x = LCD_WIDTH - 1;
rb->lcd_vline(x, BAR_HLINE_Y, BAR_HLINE_Y2);
}
#if LCD_DEPTH > 1
rb->lcd_set_foreground(front_color);
#endif
rb->lcd_putsxyf(lbl_x_minus_50 ,bar_grad_y, "%d", -50);
rb->lcd_putsxyf(lbl_x_minus_20 ,bar_grad_y, "%d", -20);
rb->lcd_putsxyf(lbl_x_0 ,bar_grad_y, "%d", 0);
rb->lcd_putsxyf(lbl_x_20 ,bar_grad_y, "%d", 20);
rb->lcd_putsxyf(lbl_x_50 ,bar_grad_y, "%d", 50);
#ifdef HAVE_LCD_COLOR
rb->lcd_set_foreground(LCD_RGBPACK(255,0,0)); /* Color screens */
#elif LCD_DEPTH > 1
rb->lcd_set_foreground(LCD_DARKGRAY); /* Greyscale screens */
#endif
if (fp_gt(wrong_by_cents, FP_ZERO))
{
rb->lcd_fillrect(bar_x_0, BAR_Y,
fixed2int(fp_mul(wrong_by_cents, LCD_FACTOR)), BAR_HEIGHT);
}
else
{
rb->lcd_fillrect(bar_x_0 + fixed2int(fp_mul(wrong_by_cents,LCD_FACTOR)),
BAR_Y,
fixed2int(fp_mul(wrong_by_cents, LCD_FACTOR)) * -1,
BAR_HEIGHT);
}
}
/* Calculate how wrong the note is and draw the GUI */
static void display_frequency (fixed freq)
{
fixed ldf, mldf;
fixed lfreq, nfreq;
fixed orig_freq;
int i, note = 0;
if (fp_lt(freq, FP_LOW))
freq = FP_LOW;
/* We calculate the frequency and its log as if */
/* the reference frequency of A were 440 Hz. */
orig_freq = freq;
lfreq = fp_add(log(freq), freq_A[settings.freq_A].logratio);
freq = fp_mul(freq, freq_A[settings.freq_A].ratio);
/* This calculates a log freq offset for note A */
/* Get the frequency to within the range of our reference table, */
/* i.e. into the right octave. */
while (fp_lt(lfreq, fp_sub(notes[0].logfreq, fp_shr(LOG_D_NOTE, 1))))
lfreq = fp_add(lfreq, LOG_2);
while (fp_gte(lfreq, fp_sub(fp_add(notes[0].logfreq, LOG_2),
fp_shr(LOG_D_NOTE, 1))))
lfreq = fp_sub(lfreq, LOG_2);
mldf = LOG_D_NOTE;
for (i=0; i<12; i++)
{
ldf = fp_gt(fp_sub(lfreq,notes[i].logfreq), FP_ZERO) ?
fp_sub(lfreq,notes[i].logfreq) : fp_neg(fp_sub(lfreq,notes[i].logfreq));
if (fp_lt(ldf, mldf))
{
mldf = ldf;
note = i;
}
}
nfreq = notes[note].freq;
while (fp_gt(fp_div(nfreq, freq), D_NOTE_SQRT))
nfreq = fp_shr(nfreq, 1);
while (fp_gt(fp_div(freq, nfreq), D_NOTE_SQRT))
nfreq = fp_shl(nfreq, 1);
ldf = fp_mul(int2fixed(1200), log(fp_div(freq,nfreq)));
rb->lcd_clear_display();
draw_bar(ldf); /* The red bar */
if(fp_round(freq) != 0)
{
/* Raise the displayed pitch an octave minus key_transposition */
/* semitones, effectively lowering it. Note that the pitch */
/* displayed alongside the frequency is unaffected. */
int transposition = 12 - settings.key_transposition;
draw_note(notes[(note + transposition) % 12].name);
if(settings.display_hz)
{
#if LCD_DEPTH > 1
rb->lcd_set_foreground(front_color);
#endif
rb->lcd_putsxyf(0, HZ_Y, "%s : %d cents (%d.%02dHz)",
notes[note].name, fp_round(ldf) ,fixed2int(orig_freq),
fp_round(fp_mul(fp_frac(orig_freq),
int2fixed(DISPLAY_HZ_PRECISION))));
}
}
rb->lcd_update();
}
#ifndef SIMULATOR
/*-----------------------------------------------------------------------
* Functions for the Yin algorithm
*
* These were all adapted from the versions in Aubio v0.3.2
* Here's what the Aubio documentation has to say:
*
* This algorithm was developped by A. de Cheveigne and H. Kawahara and
* published in:
*
* de Cheveign?, A., Kawahara, H. (2002) "YIN, a fundamental frequency
* estimator for speech and music", J. Acoust. Soc. Am. 111, 1917-1930.
*
* see http://recherche.ircam.fr/equipes/pcm/pub/people/cheveign.html
-------------------------------------------------------------------------*/
/* Find the index of the minimum element of an array of floats */
static unsigned vec_min_elem(fixed *s, unsigned buflen)
{
unsigned j, pos=0.0f;
fixed tmp = s[0];
for (j=0; j < buflen; j++)
{
if(fp_gt(tmp, s[j]))
{
pos = j;
tmp = s[j];
}
}
return pos;
}
static inline fixed aubio_quadfrac(fixed s0, fixed s1, fixed s2, fixed pf)
{
/* Original floating point version: */
/* tmp = s0 + (pf/2.0f) * (pf * ( s0 - 2.0f*s1 + s2 ) -
3.0f*s0 + 4.0f*s1 - s2);*/
/* Converted to explicit operator precedence: */
/* tmp = s0 + ((pf/2.0f) * ((((pf * ((s0 - (2*s1)) + s2)) -
(3*s0)) + (4*s1)) - s2)); */
/* I made it look like this so I could easily track the precedence and */
/* make sure it matched the original expression */
/* Oy, this is when I really wish I could do C++ operator overloading */
fixed tmp = fp_add
(
s0,
fp_mul
(
fp_shr(pf, 1),
fp_sub
(
fp_add
(
fp_sub
(
fp_mul
(
pf,
fp_add
(
fp_sub
(
s0,
fp_shl(s1, 1)
),
s2
)
),
fp_mul
(
float2fixed(3.0f),
s0
)
),
fp_shl(s1, 2)
),
s2
)
)
);
return tmp;
}
#define QUADINT_STEP float2fixed(1.0f/200.0f)
static fixed ICODE_ATTR vec_quadint_min(fixed *x, unsigned bufsize, unsigned pos, unsigned span)
{
fixed res, frac, s0, s1, s2;
fixed exactpos = int2fixed(pos);
/* init resold to something big (in case x[pos+-span]<0)) */
fixed resold = FP_MAX;
if ((pos > span) && (pos < bufsize-span))
{
s0 = x[pos-span];
s1 = x[pos] ;
s2 = x[pos+span];
/* increase frac */
for (frac = float2fixed(0.0f);
fp_lt(frac, float2fixed(2.0f));
frac = fp_add(frac, QUADINT_STEP))
{
res = aubio_quadfrac(s0, s1, s2, frac);
if (fp_lt(res, resold))
{
resold = res;
}
else
{
/* exactpos += (frac-QUADINT_STEP)*span - span/2.0f; */
exactpos = fp_add(exactpos,
fp_sub(
fp_mul(
fp_sub(frac, QUADINT_STEP),
int2fixed(span)
),
int2fixed(span)
)
);
break;
}
}
}
return exactpos;
}
/* Calculate the period of the note in the
buffer using the YIN algorithm */
/* The yin pointer is just a buffer that the algorithm uses as a work
space. It needs to be half the length of the input buffer. */
static fixed ICODE_ATTR pitchyin(int16_t *input, fixed *yin)
{
fixed retval;
unsigned j,tau = 0;
int period;
unsigned yin_size = settings.sample_size / 4;
fixed tmp = FP_ZERO, tmp2 = FP_ZERO;
yin[0] = int2fixed(1);
for (tau = 1; tau < yin_size; tau++)
{
yin[tau] = FP_ZERO;
for (j = 0; j < yin_size; j++)
{
tmp = fp_sub(int2mantissa(input[2 * j]),
int2mantissa(input[2 * (j + tau)]));
yin[tau] = fp_add(yin[tau], fp_mul(tmp, tmp));
}
tmp2 = fp_add(tmp2, yin[tau]);
if(!fp_equal(tmp2, FP_ZERO))
{
yin[tau] = fp_mul(yin[tau], fp_div(int2fixed(tau), tmp2));
}
period = tau - 3;
if(tau > 4 && fp_lt(yin[period],
yin_threshold_table[settings.yin_threshold])
&& fp_lt(yin[period], yin[period+1]))
{
retval = vec_quadint_min(yin, yin_size, period, 1);
return retval;
}
}
retval = vec_quadint_min(yin, yin_size,
vec_min_elem(yin, yin_size), 1);
return retval;
/*return FP_ZERO;*/
}
/*-----------------------------------------------------------------*/
static uint32_t ICODE_ATTR buffer_magnitude(int16_t *input)
{
unsigned n;
uint64_t tally = 0;
const unsigned size = settings.sample_size;
/* Operate on only one channel of the stereo signal */
for(n = 0; n < size; n+=2)
{
int s = input[n];
tally += s * s;
}
tally /= size / 2;
/* now tally holds the average of the squares of all the samples */
/* It must be between 0 and 0x7fff^2, so it fits in 32 bits */
return (uint32_t)tally;
}
/* Stop the recording when the buffer is full */
static void recording_callback(int status, void **start, size_t *size)
{
int tail = audio_tail ^ 1;
/* Do not overrun the reader. Reuse current buffer if full. */
if (tail != audio_head)
audio_tail = tail;
/* Always record full buffer, even if not required */
*start = audio_data[tail];
*size = BUFFER_SIZE * sizeof (int16_t);
(void)status;
}
#endif /* SIMULATOR */
/* Start recording */
static void record_data(void)
{
#ifndef SIMULATOR
/* Always record full buffer, even if not required */
rb->pcm_record_data(recording_callback, audio_data[audio_tail],
BUFFER_SIZE * sizeof (int16_t));
#endif
}
/* The main program loop */
static void record_and_get_pitch(void)
{
int quit=0, button;
bool redraw = true;
/* For tracking the latency */
/*
long timer;
char debug_string[20];
*/
#ifndef SIMULATOR
fixed period;
bool waiting = false;
#else
audio_tail = 1;
#endif
backlight_ignore_timeout();
record_data();
while(!quit)
{
while (audio_head == audio_tail && !quit) /* wait for the buffer to be filled */
{
button=pluginlib_getaction(HZ/100, plugin_contexts, PLA_ARRAY_COUNT);
switch(button)
{
case PLA_EXIT:
quit=true;
break;
case PLA_CANCEL:
rb->pcm_stop_recording();
quit = main_menu();
if(!quit)
{
redraw = true;
record_data();
}
break;
}
}
if(!quit)
{
#ifndef SIMULATOR
/* Only do the heavy lifting if the volume is high enough */
if(buffer_magnitude(audio_data[audio_head]) >
sqr(settings.volume_threshold *
rb->sound_max(SOUND_MIC_GAIN)))
{
waiting = false;
redraw = false;
#ifdef HAVE_SCHEDULER_BOOSTCTRL
rb->trigger_cpu_boost();
#endif
#ifdef PLUGIN_USE_IRAM
rb->memcpy(iram_audio_data, audio_data[audio_head],
settings.sample_size * sizeof (int16_t));
#endif
/* This returns the period of the detected pitch in samples */
period = pitchyin(iram_audio_data, yin_buffer);
/* Hz = sample rate / period */
if(fp_gt(period, FP_ZERO))
{
display_frequency(fp_period2freq(period));
}
else
{
display_frequency(FP_ZERO);
}
}
else if(redraw || !waiting)
{
waiting = true;
redraw = false;
display_frequency(FP_ZERO);
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
rb->cancel_cpu_boost();
#endif
}
/* Move to next buffer if not empty (but empty *shouldn't* happen
* here). */
if (audio_head != audio_tail)
audio_head ^= 1;
#else /* SIMULATOR */
/* Display a preselected frequency */
display_frequency(int2fixed(445));
#endif
}
}
rb->pcm_close_recording();
rb->pcm_set_frequency(REC_SAMPR_DEFAULT | SAMPR_TYPE_REC);
#ifdef HAVE_SCHEDULER_BOOSTCTRL
rb->cancel_cpu_boost();
#endif
backlight_use_settings();
}
/* Init recording, tuning, and GUI */
static void init_everything(void)
{
/* Disable all talking before initializing IRAM */
rb->talk_disable(true);
load_settings();
rb->storage_sleep();
/* Stop all playback */
rb->plugin_get_audio_buffer(NULL);
/* --------- Init the audio recording ----------------- */
rb->audio_set_output_source(AUDIO_SRC_PLAYBACK);
rb->audio_set_input_source(INPUT_TYPE, SRCF_RECORDING);
/* set to maximum gain */
rb->audio_set_recording_gain(settings.record_gain,
settings.record_gain,
AUDIO_GAIN_MIC);
/* Highest C on piano is approx 4.186 kHz, so we need just over
* 8.372 kHz to pass it. */
sample_rate = rb->round_value_to_list32(9000, rb->rec_freq_sampr,
REC_NUM_FREQ, false);
sample_rate = rb->rec_freq_sampr[sample_rate];
rb->pcm_set_frequency(sample_rate | SAMPR_TYPE_REC);
rb->pcm_init_recording();
/* avoid divsion by zero */
if(settings.lowest_freq == 0)
settings.lowest_freq = period2freq(BUFFER_SIZE / 4);
/* GUI */
#if LCD_DEPTH > 1
front_color = rb->lcd_get_foreground();
#endif
rb->lcd_getstringsize("X", &font_w, &font_h);
bar_x_0 = LCD_WIDTH / 2;
lbl_x_minus_50 = 0;
lbl_x_minus_20 = (LCD_WIDTH / 2) -
fixed2int(fp_mul(LCD_FACTOR, int2fixed(20))) - font_w;
lbl_x_0 = (LCD_WIDTH - font_w) / 2;
lbl_x_20 = (LCD_WIDTH / 2) +
fixed2int(fp_mul(LCD_FACTOR, int2fixed(20))) - font_w;
lbl_x_50 = LCD_WIDTH - 2 * font_w;
bar_grad_y = BAR_Y - BAR_PADDING - font_h;
/* Put the note right between the top and bottom text elements */
note_y = ((font_h + bar_grad_y - note_bitmaps.slide_height) / 2);
rb->talk_disable(false);
}
enum plugin_status plugin_start(const void* parameter)
{
(void)parameter;
init_everything();
record_and_get_pitch();
save_settings();
return PLUGIN_OK;
}
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