1 /**************************************************************************
3 * Open \______ \ ____ ____ | | _\_ |__ _______ ___
4 * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
5 * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
6 * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
10 * Copyright (C) 2008 Lechner Michael / smoking gnu
12 * All files in this archive are subject to the GNU General Public License.
13 * See the file COPYING in the source tree root for full license agreement.
15 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
16 * KIND, either express or implied.
18 * ----------------------------------------------------------------------------
21 * OK, this is an attempt to write an instrument tuner for rockbox.
22 * It uses a Schmitt trigger algorithm, which I copied from
23 * tuneit [ (c) 2004 Mario Lang <mlang@delysid.org> ], for detecting the
24 * fundamental freqency of a sound. A FFT algorithm would be more accurate
25 * but also much slower.
28 * - Adapt the Yin FFT algorithm, which would reduce complexity from O(n^2)
29 * to O(nlogn), theoretically reducing latency by a factor of ~10. -David
32 * 08.03.2008 Started coding
33 * 21.03.2008 Pitch detection works more or less
34 * Button definitions for most targets added
35 * 02.04.2008 Proper GUI added
36 * Todo, Major Changes and Current Limitations added
37 * 08.19.2009 Brought the code up to date with current plugin standards
38 * Made it work more nicely with color, BW and grayscale
39 * Changed pitch detection to use the Yin algorithm (better
40 * detection, but slower -- would be ~4x faster with
41 * fixed point math, I think). Code was poached from the
42 * Aubio sound processing library (aubio.org). -David
43 * 08.31.2009 Lots of changes:
44 * Added a menu to tweak settings
45 * Converted everything to fixed point (greatly improving
47 * Improved the display
48 * Improved efficiency with judicious use of cpu_boost, the
49 * backlight, and volume detection to limit unneeded
51 * Fixed a problem that caused an octave-off error
53 * 05.14.2010 Multibuffer continuous recording with two buffers
56 * CURRENT LIMITATIONS:
57 * - No gapless recording. Strictly speaking true gappless isn't possible,
58 * since the algorithm takes longer to calculate than the length of the
59 * sample, but latency could be improved a bit with proper use of the DMA
60 * recording functions.
61 * - Due to how the Yin algorithm works, latency is higher for lower
66 #include "lib/pluginlib_actions.h"
67 #include "lib/picture.h"
68 #include "lib/helper.h"
69 #include "pluginbitmaps/pitch_notes.h"
74 /* Some fixed point calculation stuff */
75 typedef int32_t fixed_data
;
80 typedef struct _fixed fixed
;
81 #define FIXED_PRECISION 18
82 #define FP_MAX ((fixed) {0x7fffffff})
83 #define FP_MIN ((fixed) {-0x80000000})
84 #define int2fixed(x) ((fixed){(x) << FIXED_PRECISION})
85 #define int2mantissa(x) ((fixed){x})
86 #define fixed2int(x) ((int)((x).a >> FIXED_PRECISION))
87 #define fixed2float(x) (((float)(x).a) / ((float)(1 << FIXED_PRECISION)))
88 #define float2fixed(x) \
89 ((fixed){(fixed_data)(x * (float)(1 << FIXED_PRECISION))})
90 /* I adapted these ones from the Rockbox fixed point library */
91 #define fp_mul(x, y) \
92 ((fixed){(((int64_t)((x).a)) * ((int64_t)((y).a))) >> (FIXED_PRECISION)})
93 #define fp_div(x, y) \
94 ((fixed){(((int64_t)((x).a)) << (FIXED_PRECISION)) / ((int64_t)((y).a))})
95 /* Operators for fixed point */
96 #define fp_add(x, y) ((fixed){(x).a + (y).a})
97 #define fp_sub(x, y) ((fixed){(x).a - (y).a})
98 #define fp_shl(x, y) ((fixed){(x).a << y})
99 #define fp_shr(x, y) ((fixed){(x).a >> y})
100 #define fp_neg(x) ((fixed){-(x).a})
101 #define fp_gt(x, y) ((x).a > (y).a)
102 #define fp_gte(x, y) ((x).a >= (y).a)
103 #define fp_lt(x, y) ((x).a < (y).a)
104 #define fp_lte(x, y) ((x).a <= (y).a)
105 #define fp_sqr(x) fp_mul((x), (x))
106 #define fp_equal(x, y) ((x).a == (y).a)
107 #define fp_round(x) (fixed2int(fp_add((x), float2fixed(0.5))))
108 #define fp_data(x) ((x).a)
109 #define fp_frac(x) (fp_sub((x), int2fixed(fixed2int(x))))
110 #define FP_ZERO ((fixed){0})
111 #define FP_LOW ((fixed){2})
113 /* Some defines for converting between period and frequency */
115 /* I introduce some divisors in this because the fixed point */
116 /* variables aren't big enough to hold higher than a certain */
117 /* value. This loses a bit of precision but it means we */
118 /* don't have to use 32.32 variables (yikes). */
119 /* With an 18-bit decimal precision, the max value in the */
120 /* integer part is 8192. Divide 44100 by 7 and it'll fit in */
122 #define fp_period2freq(x) fp_div(int2fixed(sample_rate / 7), \
123 fp_div((x),int2fixed(7)))
124 #define fp_freq2period(x) fp_period2freq(x)
125 #define period2freq(x) (sample_rate / (x))
126 #define freq2period(x) period2freq(x)
128 #define sqr(x) ((x)*(x))
130 /* Some constants for tuning */
131 #define A_FREQ float2fixed(440.0f)
132 #define D_NOTE float2fixed(1.059463094359f)
133 #define LOG_D_NOTE float2fixed(1.0f/12.0f)
134 #define D_NOTE_SQRT float2fixed(1.029302236643f)
135 #define LOG_2 float2fixed(1.0f)
137 /* The recording buffer size */
138 /* This is how much is sampled at a time. */
139 /* It also determines latency -- if BUFFER_SIZE == sample_rate then */
140 /* there'll be one sample per second, or a latency of one second. */
141 /* Furthermore, the lowest detectable frequency will be about twice */
142 /* the number of reads per second */
143 /* If we ever switch to Yin FFT algorithm then this needs to be
145 #define BUFFER_SIZE 4096
146 #define SAMPLE_SIZE 4096
147 #define SAMPLE_SIZE_MIN 1024
148 #define YIN_BUFFER_SIZE (BUFFER_SIZE / 4)
150 #define LCD_FACTOR (fp_div(int2fixed(LCD_WIDTH), int2fixed(100)))
151 /* The threshold for the YIN algorithm */
152 #define DEFAULT_YIN_THRESHOLD 5 /* 0.10 */
153 const fixed yin_threshold_table
[] IDATA_ATTR
=
171 /* Structure for the reference frequency (frequency of A)
172 * It's used for scaling the frequency before finding out
173 * the note. The frequency is scaled in a way that the main
174 * algorithm can assume the frequency of A to be 440 Hz.
178 const int frequency
; /* Frequency in Hz */
179 const fixed ratio
; /* 440/frequency */
180 const fixed logratio
; /* log2(factor) */
183 const struct freq_A_entry freq_A
[] =
185 {435, float2fixed(1.011363636), float2fixed( 0.016301812)},
186 {436, float2fixed(1.009090909), float2fixed( 0.013056153)},
187 {437, float2fixed(1.006818182), float2fixed( 0.009803175)},
188 {438, float2fixed(1.004545455), float2fixed( 0.006542846)},
189 {439, float2fixed(1.002272727), float2fixed( 0.003275132)},
190 {440, float2fixed(1.000000000), float2fixed( 0.000000000)},
191 {441, float2fixed(0.997727273), float2fixed(-0.003282584)},
192 {442, float2fixed(0.995454545), float2fixed(-0.006572654)},
193 {443, float2fixed(0.993181818), float2fixed(-0.009870244)},
194 {444, float2fixed(0.990909091), float2fixed(-0.013175389)},
195 {445, float2fixed(0.988636364), float2fixed(-0.016488123)},
198 /* Index of the entry for 440 Hz in the table (default frequency for A) */
199 #define DEFAULT_FREQ_A 5
200 #define NUM_FREQ_A (sizeof(freq_A)/sizeof(freq_A[0]))
202 /* How loud the audio has to be to start displaying pitch */
203 /* Must be between 0 and 100 */
204 #define VOLUME_THRESHOLD (50)
206 /* Change to AUDIO_SRC_LINEIN if you want to record from line-in */
208 #define INPUT_TYPE AUDIO_SRC_MIC
210 #define INPUT_TYPE AUDIO_SRC_LINEIN
213 /* How many decimal places to display for the Hz value */
214 #define DISPLAY_HZ_PRECISION 100
216 /* Where to put the various GUI elements */
219 #define LCD_RES_MIN (LCD_HEIGHT < LCD_WIDTH ? LCD_HEIGHT : LCD_WIDTH)
220 #define BAR_PADDING (LCD_RES_MIN / 32)
221 #define BAR_Y (LCD_HEIGHT * 3 / 4)
222 #define BAR_HEIGHT (LCD_RES_MIN / 4 - BAR_PADDING)
223 #define BAR_HLINE_Y (BAR_Y - BAR_PADDING)
224 #define BAR_HLINE_Y2 (BAR_Y + BAR_HEIGHT + BAR_PADDING - 1)
226 #define GRADUATION 10 /* Subdivisions of the whole 100-cent scale */
228 /* Bitmaps for drawing the note names. These need to have height
229 <= (bar_grad_y - note_y), or 15/32 * LCD_HEIGHT
231 #define NUM_NOTE_IMAGES 9
232 #define NOTE_INDEX_A 0
233 #define NOTE_INDEX_B 1
234 #define NOTE_INDEX_C 2
235 #define NOTE_INDEX_D 3
236 #define NOTE_INDEX_E 4
237 #define NOTE_INDEX_F 5
238 #define NOTE_INDEX_G 6
239 #define NOTE_INDEX_SHARP 7
240 #define NOTE_INDEX_FLAT 8
241 const struct picture note_bitmaps
=
244 BMPWIDTH_pitch_notes
,
245 BMPHEIGHT_pitch_notes
,
246 BMPHEIGHT_pitch_notes
/NUM_NOTE_IMAGES
250 static unsigned int sample_rate
;
251 static int audio_head
= 0; /* which of the two buffers to use? */
252 static volatile int audio_tail
= 0; /* which of the two buffers to record? */
253 /* It's stereo, so make the buffer twice as big */
255 static int16_t audio_data
[2][BUFFER_SIZE
] __attribute__((aligned(CACHEALIGN_SIZE
)));
256 static fixed yin_buffer
[YIN_BUFFER_SIZE
] IBSS_ATTR
;
257 #ifdef PLUGIN_USE_IRAM
258 static int16_t iram_audio_data
[BUFFER_SIZE
] IBSS_ATTR
;
260 #define iram_audio_data audio_data[audio_head]
264 /* Description of a note of scale */
267 const char *name
; /* Name of the note, e.g. "A#" */
268 const fixed freq
; /* Note frequency, Hz */
269 const fixed logfreq
; /* log2(frequency) */
272 /* Notes within one (reference) scale */
273 static const struct note_entry notes
[] =
275 {"A" , float2fixed(440.0000000f
), float2fixed(8.781359714f
)},
276 {"A#", float2fixed(466.1637615f
), float2fixed(8.864693047f
)},
277 {"B" , float2fixed(493.8833013f
), float2fixed(8.948026380f
)},
278 {"C" , float2fixed(523.2511306f
), float2fixed(9.031359714f
)},
279 {"C#", float2fixed(554.3652620f
), float2fixed(9.114693047f
)},
280 {"D" , float2fixed(587.3295358f
), float2fixed(9.198026380f
)},
281 {"D#", float2fixed(622.2539674f
), float2fixed(9.281359714f
)},
282 {"E" , float2fixed(659.2551138f
), float2fixed(9.364693047f
)},
283 {"F" , float2fixed(698.4564629f
), float2fixed(9.448026380f
)},
284 {"F#", float2fixed(739.9888454f
), float2fixed(9.531359714f
)},
285 {"G" , float2fixed(783.9908720f
), float2fixed(9.614693047f
)},
286 {"G#", float2fixed(830.6093952f
), float2fixed(9.698026380f
)},
291 static unsigned front_color
;
293 static int font_w
,font_h
;
295 static int lbl_x_minus_50
, lbl_x_minus_20
, lbl_x_0
, lbl_x_20
, lbl_x_50
;
297 /* Settings for the plugin */
298 struct tuner_settings
300 unsigned volume_threshold
;
301 unsigned record_gain
;
302 unsigned sample_size
;
303 unsigned lowest_freq
;
304 unsigned yin_threshold
;
305 int freq_A
; /* Index of the frequency of A */
310 /*=================================================================*/
311 /* Settings loading and saving(adapted from the clock plugin) */
312 /*=================================================================*/
314 #define SETTINGS_FILENAME PLUGIN_APPS_DIR "/.pitch_settings"
326 enum settings_file_status
332 /* The settings as they exist on the hard disk, so that
333 * we can know at saving time if changes have been made */
334 struct tuner_settings hdd_tuner_settings
;
336 /*---------------------------------------------------------------------*/
338 bool settings_needs_saving(struct tuner_settings
* settings
)
340 return(rb
->memcmp(settings
, &hdd_tuner_settings
, sizeof(*settings
)));
343 /*---------------------------------------------------------------------*/
345 void tuner_settings_reset(struct tuner_settings
* settings
)
347 settings
->volume_threshold
= VOLUME_THRESHOLD
;
348 settings
->record_gain
= rb
->global_settings
->rec_mic_gain
;
349 settings
->sample_size
= BUFFER_SIZE
;
350 settings
->lowest_freq
= period2freq(BUFFER_SIZE
/ 4);
351 settings
->yin_threshold
= DEFAULT_YIN_THRESHOLD
;
352 settings
->freq_A
= DEFAULT_FREQ_A
;
353 settings
->use_sharps
= true;
354 settings
->display_hz
= false;
357 /*---------------------------------------------------------------------*/
359 enum settings_file_status
tuner_settings_load(struct tuner_settings
* settings
,
362 int fd
= rb
->open(filename
, O_RDONLY
);
363 if(fd
>= 0){ /* does file exist? */
364 /* basic consistency check */
365 if(rb
->filesize(fd
) == sizeof(*settings
)){
366 rb
->read(fd
, settings
, sizeof(*settings
));
368 rb
->memcpy(&hdd_tuner_settings
, settings
, sizeof(*settings
));
372 /* Initializes the settings with default values at least */
373 tuner_settings_reset(settings
);
377 /*---------------------------------------------------------------------*/
379 enum settings_file_status
tuner_settings_save(struct tuner_settings
* settings
,
382 int fd
= rb
->creat(filename
, 0666);
383 if(fd
>= 0){ /* does file exist? */
384 rb
->write (fd
, settings
, sizeof(*settings
));
391 /*---------------------------------------------------------------------*/
393 void load_settings(void)
395 tuner_settings_load(&tuner_settings
, SETTINGS_FILENAME
);
400 /*---------------------------------------------------------------------*/
402 void save_settings(void)
404 if(!settings_needs_saving(&tuner_settings
))
407 tuner_settings_save(&tuner_settings
, SETTINGS_FILENAME
);
410 /*=================================================================*/
412 /*=================================================================*/
415 const struct button_mapping
* plugin_contexts
[]={
417 generic_increase_decrease
,
423 #define PLA_ARRAY_COUNT sizeof(plugin_contexts)/sizeof(plugin_contexts[0])
427 /* This has to match yin_threshold_table */
428 static const struct opt_items yin_threshold_text
[] =
446 static const struct opt_items accidental_text
[] =
452 void set_min_freq(int new_freq
)
454 tuner_settings
.sample_size
= freq2period(new_freq
) * 4;
456 /* clamp the sample size between min and max */
457 if(tuner_settings
.sample_size
<= SAMPLE_SIZE_MIN
)
458 tuner_settings
.sample_size
= SAMPLE_SIZE_MIN
;
459 else if(tuner_settings
.sample_size
>= BUFFER_SIZE
)
460 tuner_settings
.sample_size
= BUFFER_SIZE
;
462 /* sample size must be divisible by 4 - round up */
463 tuner_settings
.sample_size
= (tuner_settings
.sample_size
+ 3) & ~3;
470 bool exit_tuner
= false;
475 backlight_use_settings();
476 #ifdef HAVE_SCHEDULER_BOOSTCTRL
477 rb
->cancel_cpu_boost();
480 MENUITEM_STRINGLIST(menu
,"Tuner Settings",NULL
,
485 "Algorithm Pickiness",
487 "Display Frequency (Hz)",
488 "Frequency of A (Hz)",
494 choice
= rb
->do_menu(&menu
, &selection
, NULL
, false);
498 rb
->set_int("Volume Threshold", "%", UNIT_INT
,
499 &tuner_settings
.volume_threshold
,
500 NULL
, 5, 5, 95, NULL
);
503 rb
->set_int("Listening Volume", "%", UNIT_INT
,
504 &tuner_settings
.record_gain
,
505 NULL
, 1, rb
->sound_min(SOUND_MIC_GAIN
),
506 rb
->sound_max(SOUND_MIC_GAIN
), NULL
);
509 rb
->set_int("Lowest Frequency", "Hz", UNIT_INT
,
510 &tuner_settings
.lowest_freq
, set_min_freq
, 1,
511 /* Range depends on the size of the buffer */
512 sample_rate
/ (BUFFER_SIZE
/ 4),
513 sample_rate
/ (SAMPLE_SIZE_MIN
/ 4), NULL
);
517 "Algorithm Pickiness (Lower -> more discriminating)",
518 &tuner_settings
.yin_threshold
,
519 INT
, yin_threshold_text
,
520 sizeof(yin_threshold_text
) / sizeof(yin_threshold_text
[0]),
524 rb
->set_option("Display Accidentals As",
525 &tuner_settings
.use_sharps
,
526 BOOL
, accidental_text
, 2, NULL
);
529 rb
->set_bool("Display Frequency (Hz)",
530 &tuner_settings
.display_hz
);
533 freq_val
= freq_A
[tuner_settings
.freq_A
].frequency
;
534 rb
->set_int("Frequency of A (Hz)",
535 "Hz", UNIT_INT
, &freq_val
, NULL
,
536 1, freq_A
[0].frequency
, freq_A
[NUM_FREQ_A
-1].frequency
,
538 tuner_settings
.freq_A
= freq_val
- freq_A
[0].frequency
;
542 rb
->set_bool("Reset Tuner Settings?", &reset
);
544 tuner_settings_reset(&tuner_settings
);
552 /* Return to the tuner */
558 backlight_force_on();
562 /*=================================================================*/
564 /*=================================================================*/
566 /* Fixed-point log base 2*/
567 /* Adapted from python code at
568 http://en.wikipedia.org/wiki/Binary_logarithm#Algorithm
573 fixed fp
= int2fixed(1);
574 fixed res
= int2fixed(0);
576 if(fp_lte(x
, FP_ZERO
))
583 while(fp_lt(x
, int2fixed(1)))
585 res
= fp_sub(res
, int2fixed(1));
589 while(fp_gte(x
, int2fixed(2)))
591 res
= fp_add(res
, int2fixed(1));
595 /* Fractional part */
597 while(fp_gt(fp
, FP_ZERO
))
602 if(fp_gte(x
, int2fixed(2)))
605 res
= fp_add(res
, fp
);
612 /*=================================================================*/
614 /*=================================================================*/
616 /* The function name is pretty self-explaining ;) */
617 void print_int_xy(int x
, int y
, int v
)
621 rb
->lcd_set_foreground(front_color
);
623 rb
->snprintf(temp
,20,"%d",v
);
624 rb
->lcd_putsxy(x
,y
,temp
);
627 /* Print out the frequency etc */
628 void print_str(char* s
)
631 rb
->lcd_set_foreground(front_color
);
633 rb
->lcd_putsxy(0, HZ_Y
, s
);
636 /* What can I say? Read the function name... */
637 void print_char_xy(int x
, int y
, char c
)
644 rb
->lcd_set_foreground(front_color
);
647 rb
->lcd_putsxy(x
, y
, temp
);
650 /* Draw the note bitmap */
651 void draw_note(const char *note
)
654 int note_x
= (LCD_WIDTH
- BMPWIDTH_pitch_notes
) / 2;
655 int accidental_index
= NOTE_INDEX_SHARP
;
661 if(!(tuner_settings
.use_sharps
))
664 accidental_index
= NOTE_INDEX_FLAT
;
667 vertical_picture_draw_sprite(rb
->screens
[0],
672 note_x
= LCD_WIDTH
/ 2 - BMPWIDTH_pitch_notes
;
675 vertical_picture_draw_sprite(rb
->screens
[0], ¬e_bitmaps
, i
,
679 /* Draw the red bar and the white lines */
680 void draw_bar(fixed wrong_by_cents
)
685 #ifdef HAVE_LCD_COLOR
686 rb
->lcd_set_foreground(LCD_RGBPACK(255,255,255)); /* Color screens */
688 rb
->lcd_set_foreground(LCD_BLACK
); /* Greyscale screens */
691 rb
->lcd_hline(0,LCD_WIDTH
-1, BAR_HLINE_Y
);
692 rb
->lcd_hline(0,LCD_WIDTH
-1, BAR_HLINE_Y2
);
694 /* Draw graduation lines on the off-by readout */
695 for(n
= 0; n
<= GRADUATION
; n
++)
697 x
= (LCD_WIDTH
* n
+ GRADUATION
/ 2) / GRADUATION
;
700 rb
->lcd_vline(x
, BAR_HLINE_Y
, BAR_HLINE_Y2
);
703 print_int_xy(lbl_x_minus_50
,bar_grad_y
, -50);
704 print_int_xy(lbl_x_minus_20
,bar_grad_y
, -20);
705 print_int_xy(lbl_x_0
,bar_grad_y
, 0);
706 print_int_xy(lbl_x_20
,bar_grad_y
, 20);
707 print_int_xy(lbl_x_50
,bar_grad_y
, 50);
709 #ifdef HAVE_LCD_COLOR
710 rb
->lcd_set_foreground(LCD_RGBPACK(255,0,0)); /* Color screens */
712 rb
->lcd_set_foreground(LCD_DARKGRAY
); /* Greyscale screens */
715 if (fp_gt(wrong_by_cents
, FP_ZERO
))
717 rb
->lcd_fillrect(bar_x_0
, BAR_Y
,
718 fixed2int(fp_mul(wrong_by_cents
, LCD_FACTOR
)), BAR_HEIGHT
);
722 rb
->lcd_fillrect(bar_x_0
+ fixed2int(fp_mul(wrong_by_cents
,LCD_FACTOR
)),
724 fixed2int(fp_mul(wrong_by_cents
, LCD_FACTOR
)) * -1,
729 /* Calculate how wrong the note is and draw the GUI */
730 void display_frequency (fixed freq
)
738 if (fp_lt(freq
, FP_LOW
))
741 /* We calculate the frequency and its log as if */
742 /* the reference frequency of A were 440 Hz. */
744 lfreq
= fp_add(log(freq
), freq_A
[tuner_settings
.freq_A
].logratio
);
745 freq
= fp_mul(freq
, freq_A
[tuner_settings
.freq_A
].ratio
);
747 /* This calculates a log freq offset for note A */
748 /* Get the frequency to within the range of our reference table, */
749 /* i.e. into the right octave. */
750 while (fp_lt(lfreq
, fp_sub(notes
[0].logfreq
, fp_shr(LOG_D_NOTE
, 1))))
751 lfreq
= fp_add(lfreq
, LOG_2
);
752 while (fp_gte(lfreq
, fp_sub(fp_add(notes
[0].logfreq
, LOG_2
),
753 fp_shr(LOG_D_NOTE
, 1))))
754 lfreq
= fp_sub(lfreq
, LOG_2
);
758 ldf
= fp_gt(fp_sub(lfreq
,notes
[i
].logfreq
), FP_ZERO
) ?
759 fp_sub(lfreq
,notes
[i
].logfreq
) : fp_neg(fp_sub(lfreq
,notes
[i
].logfreq
));
760 if (fp_lt(ldf
, mldf
))
766 nfreq
= notes
[note
].freq
;
767 while (fp_gt(fp_div(nfreq
, freq
), D_NOTE_SQRT
))
768 nfreq
= fp_shr(nfreq
, 1);
770 while (fp_gt(fp_div(freq
, nfreq
), D_NOTE_SQRT
))
771 nfreq
= fp_shl(nfreq
, 1);
773 ldf
= fp_mul(int2fixed(1200), log(fp_div(freq
,nfreq
)));
775 rb
->lcd_clear_display();
776 draw_bar(ldf
); /* The red bar */
777 if(fp_round(freq
) != 0)
779 draw_note(notes
[note
].name
);
780 if(tuner_settings
.display_hz
)
782 rb
->snprintf(str_buf
,30, "%s : %d cents (%d.%02dHz)",
783 notes
[note
].name
, fp_round(ldf
) ,fixed2int(orig_freq
),
784 fp_round(fp_mul(fp_frac(orig_freq
),
785 int2fixed(DISPLAY_HZ_PRECISION
))));
792 /*-----------------------------------------------------------------------
793 * Functions for the Yin algorithm
795 * These were all adapted from the versions in Aubio v0.3.2
796 * Here's what the Aubio documentation has to say:
798 * This algorithm was developped by A. de Cheveigne and H. Kawahara and
801 * de Cheveign?, A., Kawahara, H. (2002) "YIN, a fundamental frequency
802 * estimator for speech and music", J. Acoust. Soc. Am. 111, 1917-1930.
804 * see http://recherche.ircam.fr/equipes/pcm/pub/people/cheveign.html
805 -------------------------------------------------------------------------*/
807 /* Find the index of the minimum element of an array of floats */
808 unsigned vec_min_elem(fixed
*s
, unsigned buflen
)
810 unsigned j
, pos
=0.0f
;
812 for (j
=0; j
< buflen
; j
++)
824 static inline fixed
aubio_quadfrac(fixed s0
, fixed s1
, fixed s2
, fixed pf
)
826 /* Original floating point version: */
827 /* tmp = s0 + (pf/2.0f) * (pf * ( s0 - 2.0f*s1 + s2 ) -
828 3.0f*s0 + 4.0f*s1 - s2);*/
829 /* Converted to explicit operator precedence: */
830 /* tmp = s0 + ((pf/2.0f) * ((((pf * ((s0 - (2*s1)) + s2)) -
831 (3*s0)) + (4*s1)) - s2)); */
833 /* I made it look like this so I could easily track the precedence and */
834 /* make sure it matched the original expression */
835 /* Oy, this is when I really wish I could do C++ operator overloading */
876 #define QUADINT_STEP float2fixed(1.0f/200.0f)
878 fixed ICODE_ATTR
vec_quadint_min(fixed
*x
, unsigned bufsize
, unsigned pos
, unsigned span
)
880 fixed res
, frac
, s0
, s1
, s2
;
881 fixed exactpos
= int2fixed(pos
);
882 /* init resold to something big (in case x[pos+-span]<0)) */
883 fixed resold
= FP_MAX
;
885 if ((pos
> span
) && (pos
< bufsize
-span
))
891 for (frac
= float2fixed(0.0f
);
892 fp_lt(frac
, float2fixed(2.0f
));
893 frac
= fp_add(frac
, QUADINT_STEP
))
895 res
= aubio_quadfrac(s0
, s1
, s2
, frac
);
896 if (fp_lt(res
, resold
))
902 /* exactpos += (frac-QUADINT_STEP)*span - span/2.0f; */
903 exactpos
= fp_add(exactpos
,
906 fp_sub(frac
, QUADINT_STEP
),
920 /* Calculate the period of the note in the
921 buffer using the YIN algorithm */
922 /* The yin pointer is just a buffer that the algorithm uses as a work
923 space. It needs to be half the length of the input buffer. */
925 fixed ICODE_ATTR
pitchyin(int16_t *input
, fixed
*yin
)
930 unsigned yin_size
= tuner_settings
.sample_size
/ 4;
932 fixed tmp
= FP_ZERO
, tmp2
= FP_ZERO
;
933 yin
[0] = int2fixed(1);
934 for (tau
= 1; tau
< yin_size
; tau
++)
937 for (j
= 0; j
< yin_size
; j
++)
939 tmp
= fp_sub(int2mantissa(input
[2 * j
]),
940 int2mantissa(input
[2 * (j
+ tau
)]));
941 yin
[tau
] = fp_add(yin
[tau
], fp_mul(tmp
, tmp
));
943 tmp2
= fp_add(tmp2
, yin
[tau
]);
944 if(!fp_equal(tmp2
, FP_ZERO
))
946 yin
[tau
] = fp_mul(yin
[tau
], fp_div(int2fixed(tau
), tmp2
));
949 if(tau
> 4 && fp_lt(yin
[period
],
950 yin_threshold_table
[tuner_settings
.yin_threshold
])
951 && fp_lt(yin
[period
], yin
[period
+1]))
953 retval
= vec_quadint_min(yin
, yin_size
, period
, 1);
957 retval
= vec_quadint_min(yin
, yin_size
,
958 vec_min_elem(yin
, yin_size
), 1);
963 /*-----------------------------------------------------------------*/
965 uint32_t ICODE_ATTR
buffer_magnitude(int16_t *input
)
969 const unsigned size
= tuner_settings
.sample_size
;
971 /* Operate on only one channel of the stereo signal */
972 for(n
= 0; n
< size
; n
+=2)
980 /* now tally holds the average of the squares of all the samples */
981 /* It must be between 0 and 0x7fff^2, so it fits in 32 bits */
982 return (uint32_t)tally
;
985 /* Stop the recording when the buffer is full */
987 int recording_callback(int status
)
989 int tail
= audio_tail
^ 1;
991 /* Do not overrun the reader. Reuse current buffer if full. */
992 if (tail
!= audio_head
)
995 /* Always record full buffer, even if not required */
996 rb
->pcm_record_more(audio_data
[tail
],
997 BUFFER_SIZE
* sizeof (int16_t));
1004 /* Start recording */
1005 static void record_data(void)
1008 /* Always record full buffer, even if not required */
1009 rb
->pcm_record_data(recording_callback
, audio_data
[audio_tail
],
1010 BUFFER_SIZE
* sizeof (int16_t));
1014 /* The main program loop */
1015 void record_and_get_pitch(void)
1019 /* For tracking the latency */
1022 char debug_string[20];
1026 bool waiting
= false;
1031 backlight_force_on();
1037 while (audio_head
== audio_tail
&& !quit
) /* wait for the buffer to be filled */
1039 button
=pluginlib_getaction(HZ
/100, plugin_contexts
, PLA_ARRAY_COUNT
);
1048 rb
->pcm_stop_recording();
1049 quit
= main_menu() != 0;
1064 /* Only do the heavy lifting if the volume is high enough */
1065 if(buffer_magnitude(audio_data
[audio_head
]) >
1066 sqr(tuner_settings
.volume_threshold
*
1067 rb
->sound_max(SOUND_MIC_GAIN
)))
1072 #ifdef HAVE_SCHEDULER_BOOSTCTRL
1073 rb
->trigger_cpu_boost();
1075 #ifdef PLUGIN_USE_IRAM
1076 rb
->memcpy(iram_audio_data
, audio_data
[audio_head
],
1077 tuner_settings
.sample_size
* sizeof (int16_t));
1079 /* This returns the period of the detected pitch in samples */
1080 period
= pitchyin(iram_audio_data
, yin_buffer
);
1081 /* Hz = sample rate / period */
1082 if(fp_gt(period
, FP_ZERO
))
1084 display_frequency(fp_period2freq(period
));
1088 display_frequency(FP_ZERO
);
1091 else if(redraw
|| !waiting
)
1095 display_frequency(FP_ZERO
);
1096 #ifdef HAVE_ADJUSTABLE_CPU_FREQ
1097 rb
->cancel_cpu_boost();
1101 /* Move to next buffer if not empty (but empty *shouldn't* happen
1103 if (audio_head
!= audio_tail
)
1105 #else /* SIMULATOR */
1106 /* Display a preselected frequency */
1107 display_frequency(int2fixed(445));
1111 rb
->pcm_close_recording();
1112 rb
->pcm_set_frequency(HW_SAMPR_DEFAULT
);
1113 #ifdef HAVE_SCHEDULER_BOOSTCTRL
1114 rb
->cancel_cpu_boost();
1117 backlight_use_settings();
1120 /* Init recording, tuning, and GUI */
1121 void init_everything(void)
1123 /* Disable all talking before initializing IRAM */
1124 rb
->talk_disable(true);
1126 PLUGIN_IRAM_INIT(rb
);
1130 /* Stop all playback (if no IRAM, otherwise IRAM_INIT would have) */
1131 rb
->plugin_get_audio_buffer(NULL
);
1133 /* --------- Init the audio recording ----------------- */
1134 rb
->audio_set_output_source(AUDIO_SRC_PLAYBACK
);
1135 rb
->audio_set_input_source(INPUT_TYPE
, SRCF_RECORDING
);
1137 /* set to maximum gain */
1138 rb
->audio_set_recording_gain(tuner_settings
.record_gain
,
1139 tuner_settings
.record_gain
,
1142 /* Highest C on piano is approx 4.186 kHz, so we need just over
1143 * 8.372 kHz to pass it. */
1144 sample_rate
= rb
->round_value_to_list32(9000, rb
->rec_freq_sampr
,
1145 REC_NUM_FREQ
, false);
1146 sample_rate
= rb
->rec_freq_sampr
[sample_rate
];
1147 rb
->pcm_set_frequency(sample_rate
);
1148 rb
->pcm_init_recording();
1152 front_color
= rb
->lcd_get_foreground();
1154 rb
->lcd_getstringsize("X", &font_w
, &font_h
);
1156 bar_x_0
= LCD_WIDTH
/ 2;
1158 lbl_x_minus_20
= (LCD_WIDTH
/ 2) -
1159 fixed2int(fp_mul(LCD_FACTOR
, int2fixed(20))) - font_w
;
1160 lbl_x_0
= (LCD_WIDTH
- font_w
) / 2;
1161 lbl_x_20
= (LCD_WIDTH
/ 2) +
1162 fixed2int(fp_mul(LCD_FACTOR
, int2fixed(20))) - font_w
;
1163 lbl_x_50
= LCD_WIDTH
- 2 * font_w
;
1165 bar_grad_y
= BAR_Y
- BAR_PADDING
- font_h
;
1166 /* Put the note right between the top and bottom text elements */
1167 note_y
= ((font_h
+ bar_grad_y
- note_bitmaps
.slide_height
) / 2);
1169 rb
->talk_disable(false);
1173 enum plugin_status
plugin_start(const void* parameter
) NO_PROF_ATTR
1178 record_and_get_pitch();