From 20245a2edb219df444566cda5ea0e8eaac813de7 Mon Sep 17 00:00:00 2001 From: alle Date: Tue, 8 Sep 2009 16:45:05 +0000 Subject: [PATCH] New plugin: pitch detector (FS#8768) by Michael Lechner and David Johnston git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22663 a1c6a512-1295-4272-9138-f99709370657 --- apps/plugins/CATEGORIES | 1 + apps/plugins/SOURCES | 4 + apps/plugins/pitch_detector.c | 1055 +++++++++++++++++++++++++++++++++++++++++ docs/CREDITS | 1 + 4 files changed, 1061 insertions(+) create mode 100644 apps/plugins/pitch_detector.c diff --git a/apps/plugins/CATEGORIES b/apps/plugins/CATEGORIES index 3dc11b636..ae764ca12 100644 --- a/apps/plugins/CATEGORIES +++ b/apps/plugins/CATEGORIES @@ -58,6 +58,7 @@ pacbox,games pdbox,viewers pegbox,games pictureflow,demos +pitch_detector,apps plasma,demos png,viewers pong,games diff --git a/apps/plugins/SOURCES b/apps/plugins/SOURCES index 95ffb1c9d..b9061ee74 100644 --- a/apps/plugins/SOURCES +++ b/apps/plugins/SOURCES @@ -29,6 +29,10 @@ firmware_flash.c rockbox_flash.c #endif /* CONFIG_CPU */ +#if (CONFIG_CODEC == SWCODEC) && defined(HAVE_RECORDING) && \ + (defined(HAVE_LINE_IN) || defined(HAVE_MIC_IN)) +pitch_detector.c +#endif #if (CONFIG_CODEC == SWCODEC) || !defined(SIMULATOR) metronome.c diff --git a/apps/plugins/pitch_detector.c b/apps/plugins/pitch_detector.c new file mode 100644 index 000000000..8d0b3b795 --- /dev/null +++ b/apps/plugins/pitch_detector.c @@ -0,0 +1,1055 @@ +/************************************************************************** + * __________ __ ___. + * 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 ], for detecting the + * fundamental freqency of a sound. A FFT algorithm would be more accurate + * but also much slower. + * + * TODO: + * - Find someone who knows how recording actually works, and rewrite the + * recording code to use proper, gapless recording with a callback function + * that provides new buffer, instead of stopping and restarting recording + * everytime the buffer is full + * - Convert all floating point operations to fixed-point + * - 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 + * + * + * 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" + +PLUGIN_HEADER + +/* First figure out what sample rate we're going to use */ +#if (REC_SAMPR_CAPS & SAMPR_CAP_44) +#define SAMPLE_RATE SAMPR_44 +#elif (REC_SAMPR_CAPS & SAMPR_CAP_22) +#define SAMPLE_RATE SAMPR_22 +#elif (REC_SAMPR_CAPS & SAMPR_CAP_11) +#define SAMPLE_RATE SAMPR_11 +#endif + +/* Some fixed point calculation stuff */ +typedef int32_t fixed_data; +struct _fixed +{ + fixed_data a; +}; +typedef struct _fixed 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).a >> FIXED_PRECISION)) +#define fixed2float(x) (((float)(x).a) / ((float)(1 << FIXED_PRECISION))) +#define float2fixed(x) \ + ((fixed){(fixed_data)(x * (float)(1 << FIXED_PRECISION))}) +/* I adapted these ones from the Rockbox fixed point library */ +#define fp_mul(x, y) \ + ((fixed){(((int64_t)((x).a)) * ((int64_t)((y).a))) >> (FIXED_PRECISION)}) +#define fp_div(x, y) \ + ((fixed){(((int64_t)((x).a)) << (FIXED_PRECISION)) / ((int64_t)((y).a))}) +/* Operators for fixed point */ +#define fp_add(x, y) ((fixed){(x).a + (y).a}) +#define fp_sub(x, y) ((fixed){(x).a - (y).a}) +#define fp_shl(x, y) ((fixed){(x).a << y}) +#define fp_shr(x, y) ((fixed){(x).a >> y}) +#define fp_neg(x) ((fixed){-(x).a}) +#define fp_gt(x, y) ((x).a > (y).a) +#define fp_gte(x, y) ((x).a >= (y).a) +#define fp_lt(x, y) ((x).a < (y).a) +#define fp_lte(x, y) ((x).a <= (y).a) +#define fp_sqr(x) fp_mul((x), (x)) +#define fp_equal(x, y) ((x).a == (y).a) +#define fp_round(x) (fixed2int(fp_add((x), float2fixed(0.5)))) +#define fp_data(x) ((x).a) +#define fp_frac(x) (fp_sub((x), int2fixed(fixed2int(x)))) +#define FP_ZERO ((fixed){0}) +#define FP_LOW ((fixed){1}) + +/* 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 */ +/* vallue. 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 YIN_THRESHOLD float2fixed(0.05f) + +/* 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 */ +#define INPUT_TYPE AUDIO_SRC_MIC + +/* How many decimal places to display for the Hz value */ +#define DISPLAY_HZ_PRECISION 100 + +typedef signed short audio_sample_type; +/* It's stereo, so make the buffer twice as big */ +audio_sample_type audio_data[BUFFER_SIZE]; +fixed yin_buffer[YIN_BUFFER_SIZE]; +static int recording=0; + +/* Frequencies of all the notes of the scale */ +static const fixed freqs[12] = +{ + float2fixed(440.0f), /* A */ + float2fixed(466.1637615f), /* A# */ + float2fixed(493.8833013f), /* etc... */ + float2fixed(523.2511306f), + float2fixed(554.3652620f), + float2fixed(587.3295358f), + float2fixed(622.2539674f), + float2fixed(659.2551138f), + float2fixed(698.4564629f), + float2fixed(739.9888454f), + float2fixed(783.9908720f), + float2fixed(830.6093952f), +}; +/* logarithm of all the notes of the scale */ +static const fixed lfreqs[12] = +{ + float2fixed(8.781359714f), + float2fixed(8.864693047f), + float2fixed(8.948026380f), + float2fixed(9.031359714f), + float2fixed(9.114693047f), + float2fixed(9.198026380f), + float2fixed(9.281359714f), + float2fixed(9.364693047f), + float2fixed(9.448026380f), + float2fixed(9.531359714f), + float2fixed(9.614693047f), + float2fixed(9.698026380f), +}; + +/* GUI */ +static unsigned back_color, front_color; +static int font_w, font_h; +static int bar_x_minus_50, bar_x_minus_20, bar_x_0, bar_x_20, bar_x_50; +static int letter_y; /* y of the notes letters */ +static int note_bar_y; /* y of the yellow bars (under the notes) */ +static int bar_h; /* height of the yellow (note) and red (error) bars */ +static int freq_y; /* y of the line with the frequency */ +static int error_ticks_y; /* y of the error values (-50, -20, ...) */ +static int error_hline_y; /* y of the top line for error bar */ +static int error_hline_y2;/* y of the bottom line for error bar */ +static int error_bar_y; /* y of the error bar */ +static int error_bar_margin; /* distance between the error bar and the hline */ + +static const char *english_notes[12] = {"A","A#","B","C","C#","D","D#","E", + "F","F#", "G", "G#"}; +static const char gui_letters[8] = {'A', 'B', 'C', 'D', 'E', 'F', 'G', '#'}; +static const char **notes = english_notes; + +/*Settings for the plugin */ + +struct tuner_settings +{ + unsigned volume_threshold; + unsigned record_gain; + unsigned sample_size; + unsigned lowest_freq; + unsigned yin_threshold; +} tuner_settings; + +/*=================================================================*/ +/* MENU */ +/*=================================================================*/ + +/* Keymaps */ +const struct button_mapping* plugin_contexts[]={ + generic_actions, + generic_increase_decrease, + generic_directions, +#if NB_SCREENS == 2 + remote_directions +#endif +}; +#define PLA_ARRAY_COUNT sizeof(plugin_contexts)/sizeof(plugin_contexts[0]) + +fixed yin_threshold_table[] = +{ + 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), +}; + +#define DEFAULT_YIN_THRESHOLD 5 /* 0.10 */ + +/* Option strings */ +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 }, +}; + +void set_min_freq(int new_freq) +{ + tuner_settings.sample_size = freq2period(new_freq) * 4; + + /* clamp the sample size between min and max */ + if(tuner_settings.sample_size <= SAMPLE_SIZE_MIN) + tuner_settings.sample_size = SAMPLE_SIZE_MIN; + else if(tuner_settings.sample_size >= BUFFER_SIZE) + tuner_settings.sample_size = BUFFER_SIZE; + /* sample size must be divisible by 4 */ + else if(tuner_settings.sample_size % 4 != 0) + tuner_settings.sample_size += 4 - (tuner_settings.sample_size % 4); +} + +bool main_menu(void) +{ + int selection=0; + bool done = false; + bool exit_tuner=false; + int choice; + + MENUITEM_STRINGLIST(menu,"Tuner Settings",NULL, + "Return to Tuner", + "Volume Threshold", + "Listening Volume", + "Lowest Frequency", + "Algorithm Pickiness", + "Quit"); + + while(!done) + { + choice = rb->do_menu(&menu, &selection, NULL, false); + switch(choice) + { + case 1: + rb->set_int("Volume Threshold", "%", UNIT_INT, + &tuner_settings.volume_threshold, + NULL, 5, 5, 95, NULL); + break; + case 2: + rb->set_int("Listening Volume", "%", UNIT_INT, + &tuner_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, + &tuner_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)", + &tuner_settings.yin_threshold, + INT, yin_threshold_text, + sizeof(yin_threshold_text) / + sizeof(yin_threshold_text[0]), + NULL); + break; + case 5: + exit_tuner = true; + done = true; + break; + case 0: + default: /* Return to the tuner */ + done = true; + break; + + } + } + return(exit_tuner); +} + +/*=================================================================*/ +/* Settings loading and saving(adapted from the clock plugin) */ +/*=================================================================*/ + +#define SETTINGS_FILENAME PLUGIN_APPS_DIR "/.pitch_settings" + +enum message +{ + MESSAGE_LOADING, + MESSAGE_LOADED, + MESSAGE_ERRLOAD, + MESSAGE_SAVING, + MESSAGE_SAVED, + MESSAGE_ERRSAVE +}; + +enum settings_file_status +{ + LOADED, ERRLOAD, + SAVED, ERRSAVE +}; + +/* The settings as they exist on the hard disk, so that + * we can know at saving time if changes have been made */ +struct tuner_settings hdd_tuner_settings; + +/*---------------------------------------------------------------------*/ + +bool settings_needs_saving(struct tuner_settings* settings) +{ + return(rb->memcmp(settings, &hdd_tuner_settings, sizeof(*settings))); +} + +/*---------------------------------------------------------------------*/ + +void tuner_settings_reset(struct tuner_settings* settings) +{ + settings->volume_threshold = VOLUME_THRESHOLD; + settings->record_gain = rb->global_settings->rec_mic_gain; + settings->sample_size = BUFFER_SIZE; + settings->lowest_freq = period2freq(BUFFER_SIZE / 4); + settings->yin_threshold = DEFAULT_YIN_THRESHOLD; +} + +/*---------------------------------------------------------------------*/ + +enum settings_file_status tuner_settings_load(struct tuner_settings* settings, + char* filename) +{ + int fd = rb->open(filename, O_RDONLY); + if(fd >= 0){ /* does file exist? */ + /* basic consistency check */ + if(rb->filesize(fd) == sizeof(*settings)){ + rb->read(fd, settings, sizeof(*settings)); + rb->close(fd); + rb->memcpy(&hdd_tuner_settings, settings, sizeof(*settings)); + return(LOADED); + } + } + /* Initializes the settings with default values at least */ + tuner_settings_reset(settings); + return(ERRLOAD); +} + +/*---------------------------------------------------------------------*/ + +enum settings_file_status tuner_settings_save(struct tuner_settings* settings, + char* filename) +{ + int fd = rb->creat(filename); + if(fd >= 0){ /* does file exist? */ + rb->write (fd, settings, sizeof(*settings)); + rb->close(fd); + return(SAVED); + } + return(ERRSAVE); +} + +/*---------------------------------------------------------------------*/ + +void load_settings(void) +{ + tuner_settings_load(&tuner_settings, SETTINGS_FILENAME); + + rb->storage_sleep(); +} + +/*---------------------------------------------------------------------*/ + +void save_settings(void) +{ + if(!settings_needs_saving(&tuner_settings)) + return; + + tuner_settings_save(&tuner_settings, SETTINGS_FILENAME); +} + +/*=================================================================*/ +/* Binary Log */ +/*=================================================================*/ + +/* Fixed-point log base 2*/ +/* Adapted from python code at + http://en.wikipedia.org/wiki/Binary_logarithm#Algorithm +*/ +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 */ +/*=================================================================*/ + +/* The function name is pretty self-explaining ;) */ +void print_int_xy(int x, int y, int v) +{ + char temp[20]; + + rb->lcd_set_foreground(front_color); + rb->snprintf(temp,20,"%d",v); + rb->lcd_putsxy(x,y,temp); + rb->lcd_update(); +} + +/* Print out the frequency etc - will be removed later on */ +void print_str(char* s) +{ + rb->lcd_set_foreground(front_color); + rb->lcd_putsxy(0, freq_y, s); + rb->lcd_update(); +} + +/* What can I say? Read the function name... */ +void print_char_xy(int x, int y, char c) +{ + char temp[2]; + + temp[0]=c; + temp[1]=0; + rb->lcd_set_foreground(front_color); + + rb->lcd_putsxy(x, y, temp); +} + +/* Draw the red bar and the white lines */ +void draw_bar(fixed wrong_by_cents) +{ +#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 */ +#else + rb->lcd_set_foreground(LCD_BLACK); /* Black and white screens */ +#endif + + if (fp_gt(wrong_by_cents, FP_ZERO)) + { + rb->lcd_fillrect(bar_x_0, error_bar_y, + fixed2int(fp_mul(wrong_by_cents, LCD_FACTOR)), bar_h); + } + else + { + rb->lcd_fillrect(bar_x_0 + fixed2int(fp_mul(wrong_by_cents,LCD_FACTOR)), + error_bar_y, + fixed2int(fp_mul(wrong_by_cents, LCD_FACTOR)) * -1, + bar_h); + } +#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 */ +#else + rb->lcd_set_foreground(LCD_BLACK); /* Black and white screens */ +#endif + + rb->lcd_hline(0,LCD_WIDTH-1, error_hline_y); + rb->lcd_hline(0,LCD_WIDTH-1, error_hline_y2); + rb->lcd_vline(LCD_WIDTH / 2, error_hline_y, error_hline_y2); + + print_int_xy(bar_x_minus_50 , error_ticks_y, -50); + print_int_xy(bar_x_minus_20 , error_ticks_y, -20); + print_int_xy(bar_x_0 , error_ticks_y, 0); + print_int_xy(bar_x_20 , error_ticks_y, 20); + print_int_xy(bar_x_50 , error_ticks_y, 50); +} + +/* Print the letters A-G and the # on the screen */ +void draw_letters(void) +{ + int i; + + rb->lcd_set_foreground(front_color); + + for (i=0; i<8; i++) + { + print_char_xy(i*(LCD_WIDTH / 8 ) + font_w, letter_y, gui_letters[i]); + } +} + +/* Draw the yellow point(s) below the letters and the '#' */ +void draw_points(const char *s) +{ + int i; + + i = s[0]-'A'; +#ifdef HAVE_LCD_COLOR + rb->lcd_set_foreground(LCD_RGBPACK(255,255,0)); /* Color screens */ +#elif LCD_DEPTH > 1 + rb->lcd_set_foreground(LCD_DARKGRAY); /* Grey screens */ +#else + rb->lcd_set_foreground(LCD_BLACK); /* Black and White screens */ +#endif + + rb->lcd_fillrect(i*(LCD_WIDTH / 8 ) + font_w, note_bar_y, font_w, font_h); + + if (s[1] == '#') + rb->lcd_fillrect(7*(LCD_WIDTH / 8 ) + font_w, note_bar_y, font_w, font_h); +} + +/* Calculate how wrong the note is and draw the GUI */ +void display_frequency (fixed freq) +{ + fixed ldf, mldf; + fixed lfreq, nfreq; + int i, note = 0; + bool draw_freq; + char str_buf[30]; + + if (fp_lt(freq, FP_LOW)) + freq = FP_LOW; + lfreq = log(freq); + + /* Get the frequency to within the range of our reference table */ + while (fp_lt(lfreq, fp_sub(lfreqs[0], fp_shr(LOG_D_NOTE, 1)))) + lfreq = fp_add(lfreq, LOG_2); + while (fp_gte(lfreq, fp_sub(fp_add(lfreqs[0], 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,lfreqs[i]), FP_ZERO) ? + fp_sub(lfreq,lfreqs[i]) : fp_neg(fp_sub(lfreq,lfreqs[i])); + if (fp_lt(ldf, mldf)) + { + mldf = ldf; + note = i; + } + } + nfreq = freqs[note]; + 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))); + + draw_freq = (fp_round(freq) != 0); + if (draw_freq) + { + rb->snprintf(str_buf,30, "%s : %d cents (%d.%dHz)", + notes[note], fp_round(ldf) ,fixed2int(freq), + fp_round(fp_mul(fp_frac(freq), + int2fixed(DISPLAY_HZ_PRECISION)))); + } + else + { + ldf = FP_ZERO; /* prevents red bar at -32 cents when freq= 0*/ + } + + rb->lcd_clear_display(); + draw_bar(ldf); /* The red bar */ + draw_letters(); /* The A-G letters and the # */ + if (draw_freq) + { + draw_points(notes[note]); /* The yellow point(s) */ + print_str(str_buf); + } + rb->lcd_update(); +} + +/*----------------------------------------------------------------------- + * 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 */ +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; +} + + +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) + +fixed 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. */ + +fixed pitchyin(audio_sample_type *input, fixed *yin) +{ + fixed retval; + unsigned j,tau = 0; + int period; + unsigned yin_size = tuner_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[tuner_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;*/ +} + +/*-----------------------------------------------------------------*/ + +uint32_t buffer_magnitude(audio_sample_type *input) +{ + unsigned n; + uint64_t tally = 0; + + /* Operate on only one channel of the stereo signal */ + for(n = 0; n < tuner_settings.sample_size; n+=2) + { + tally += (uint64_t)input[n] * (uint64_t)input[n]; + } + + tally /= tuner_settings.sample_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 */ +int recording_callback(int status) +{ + (void) status; + + rb->pcm_stop_recording(); + recording=0; + return -1; +} + +/* The main program loop */ +void record_and_get_pitch(void) +{ + int quit=0, button; + bool redraw = true; + /* For tracking the latency */ + /* + long timer; + char debug_string[20]; + */ + fixed period; + bool waiting = false; + + while(!quit) + { + /* Start recording */ + rb->pcm_record_data(recording_callback, (void *) audio_data, + (size_t) tuner_settings.sample_size * + sizeof(audio_sample_type)); + recording=1; + + while (recording && !quit) /* wait for the buffer to be filled */ + { + rb->yield(); + button=pluginlib_getaction(0, plugin_contexts, PLA_ARRAY_COUNT); + switch(button) + { + case PLA_QUIT: + quit=true; + rb->yield(); + break; + + case PLA_MENU: + if(main_menu()) + quit=true; + rb->yield(); + redraw = true; + break; + + default: + rb->yield(); + + break; + } + } + + /* Let's keep track of how long this takes */ + /* timer = *(rb->current_tick); */ + + if(!quit) + { + /* Only do the heavy lifting if the volume is high enough */ + if(buffer_magnitude(audio_data) > + sqr(tuner_settings.volume_threshold * + rb->sound_max(SOUND_MIC_GAIN))) + { + if(waiting) + { + #ifdef HAVE_ADJUSTABLE_CPU_FREQ + rb->cpu_boost(true); + #endif + waiting = false; + } + + rb->backlight_on(); + redraw = false; + + /* This returns the period of the detected pitch in samples */ + period = pitchyin(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; + #ifdef HAVE_ADJUSTABLE_CPU_FREQ + rb->cpu_boost(false); + #endif + /*rb->backlight_off();*/ + display_frequency(FP_ZERO); + } + + /*rb->snprintf(debug_string, 20, "%x,%x", + audio_data[50], audio_data[52]); + rb->lcd_putsxy(0, 40, debug_string); + rb->lcd_update(); + */ + + /* Print out how long it took to find the pitch */ + /* + rb->snprintf(debug_string, 20, "latency: %ld", + *(rb->current_tick) - timer); + rb->lcd_putsxy(0, 40, debug_string); + rb->lcd_update(); + */ + } + } + rb->pcm_close_recording(); +#ifdef HAVE_ADJUSTABLE_CPU_FREQ + rb->cpu_boost(false); +#endif +} + +/* Init recording, tuning, and GUI */ +void init_everything(void) +{ + load_settings(); + + /* --------- 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(tuner_settings.record_gain, + tuner_settings.record_gain, + AUDIO_GAIN_MIC); + + rb->pcm_set_frequency(SAMPLE_RATE); + rb->pcm_apply_settings(); + + rb->pcm_init_recording(); + + /* GUI */ + back_color = rb->lcd_get_background(); + front_color = rb->lcd_get_foreground(); + rb->lcd_getstringsize("X", &font_w, &font_h); + + bar_x_minus_50 = 0; + bar_x_minus_20 = (LCD_WIDTH / 2) - + fixed2int(fp_mul(LCD_FACTOR, int2fixed(20))) - font_w; + bar_x_0 = LCD_WIDTH / 2; + bar_x_20 = (LCD_WIDTH / 2) + + fixed2int(fp_mul(LCD_FACTOR, int2fixed(20))) - font_w; + bar_x_50 = LCD_WIDTH - 2 * font_w; + + letter_y = 10; + note_bar_y = letter_y + font_h; + bar_h = font_h; + freq_y = note_bar_y + bar_h + 3; + error_ticks_y = freq_y + font_h + 8; + error_hline_y = error_ticks_y + font_h + 2; + error_bar_margin = 2; + error_bar_y = error_hline_y + error_bar_margin; + error_hline_y2 = error_bar_y + bar_h + error_bar_margin; +} + + +enum plugin_status plugin_start(const void* parameter) NO_PROF_ATTR +{ + (void)parameter; + + init_everything(); + record_and_get_pitch(); + save_settings(); + + return 0; +} diff --git a/docs/CREDITS b/docs/CREDITS index 46a555b68..906677824 100644 --- a/docs/CREDITS +++ b/docs/CREDITS @@ -492,6 +492,7 @@ Johannes Boy Jason Yu Aaron DeMille Tomasz Kowalczyk +Michael Lechner The libmad team The wavpack team -- 2.11.4.GIT