diff options
Diffstat (limited to 'util/spkmodem_decode/spkmodem-decode.c')
| -rw-r--r-- | util/spkmodem_decode/spkmodem-decode.c | 725 |
1 files changed, 725 insertions, 0 deletions
diff --git a/util/spkmodem_decode/spkmodem-decode.c b/util/spkmodem_decode/spkmodem-decode.c new file mode 100644 index 00000000..3b3b33f8 --- /dev/null +++ b/util/spkmodem_decode/spkmodem-decode.c @@ -0,0 +1,725 @@ +/* + * SPDX-License-Identifier: GPL-2.0-or-later + * Copyright (c) 2013 Free Software Foundation, Inc. + * Copyright (c) 2023, 2026 Leah Rowe <leah@libreboot.org> + * + * This program receives text encoded as pulses on the PC speaker, + * and decodes them via simple FSK (Frequency Shift Keying) + * demodulation and FIR (Finite Impulse Response) frequency + * discriminator. + * + * It waits for specific tones at specific intervals. + * It detects tones within the audio stream and reconstructs + * characters bit-by-bit as the encoded modem signal is received. + * This is performance-efficient on most CPUs, and has relatively + * high tolerance for noisy signals (similar to techniques used + * for data stored on audio cassette tapes). + * + * This is a special interface provided by coreboot and GNU GRUB, + * for computers that lack serial ports (useful for debugging). + * Note that GRUB and coreboot can both send these signals; this + * tool merely decodes them. This tool does not *encode*, only + * decode. + * + * Usage example (NOTE: little endian!): + * parec --channels=1 --rate=48000 --format=s16le | ./spkmodem-decode + * + * Originally provided by GNU GRUB, this version is a heavily + * modified fork that complies with the OpenBSD Kernel Source + * File Style Guide (KNF) instead of GNU coding standards; it + * emphasises strict error handling, portability and code + * quality, as characterised by OpenBSD projects. Several magic + * numbers have been tidied up, calculated (not hardcoded) and + * thoroughly explained, unlike in the original version. + * + * The original version was essentially a blob, masquerading as + * source code. This forked source code is therefore the result + * of extensive reverse engineering (of the GNU source code)! + * This cleaned up code and extensive commenting will thoroughly + * explain how the decoding works. This was done as an academic + * exercise in 2023, continuing in 2026. + * + * This fork of spkmodem-recv, called spkmodem-decode, is provided + * with Libreboot releases: + * https://libreboot.org/ + * + * The original GNU version is here, if you're morbidly curious: + * https://cgit.git.savannah.gnu.org/cgit/grub.git/plain/util/spkmodem-recv.c?id=3dce38eb196f47bdf86ab028de74be40e13f19fd + * + * Libreboot's version was renamed to spkmodem-decode on 12 March 2026, + * since Libreboot's version no longer closely resembles the GNU + * version at all; ergo, a full rename was in order. GNU's version + * was called spkmodem-recv. + */ + +#define _POSIX_SOURCE + +/* + * For OpenBSD define, to detect version + * for deciding whether to use pledge(2) + */ +#ifdef __OpenBSD__ +#include <sys/param.h> +#endif + +#include <errno.h> +#include <limits.h> +#include <stdio.h> +#include <stdarg.h> +#include <stdlib.h> +#include <string.h> +#include <unistd.h> + +/* + * spkmodem is essentially using FSK (Frequency Shift Keying) + * with two primary tones representing encoded bits, + * separated by a framing tone. + * Very cheap on CPU cycles and avoids needing something more + * expensive like FFT or Goertzel filters, and tolerates + * weak/noisy signals. + */ + +/* + * Frequency of audio in Hz + * WARNING: if changing, make sure to adjust + * SAMPLES_PER_FRAME accordingly (see maths below) + */ +#define SAMPLE_RATE 48000 + +/* + * One analysis frame spans 5 ms. + * + * frame_time = SAMPLES_PER_FRAME / SAMPLE_RATE + * + * With the default sample rate (48 kHz): + * + * frame_time = N / 48000 + * 0.005 s = N / 48000 + * N = 0.005 × 48000 = 240 samples + */ +#define SAMPLES_PER_FRAME 240 + +/* + * Number of analysis frames per second. + * + * Each increment in the frequency counters corresponds + * roughly to this many Hertz of tone frequency. + * + * With the default values: + * FRAME_RATE = 48000 / 240 = 200 Hz + */ +#define FRAME_RATE ((SAMPLE_RATE) / (SAMPLES_PER_FRAME)) + +/* + * Two FIR windows are maintained; one for data tone, + * and one for the separator tone. They are positioned + * one frame apart in the ring buffer. + */ +#define MAX_SAMPLES (2 * (SAMPLES_PER_FRAME)) + +/* + * Approx byte offset for ring buffer span, just for + * easier debug output correlating to the audio stream. + */ +#define SAMPLE_OFFSET ((MAX_SAMPLES) * (sizeof(short))) + +/* + * Expected tone ranges (approximate, derived from spkmodem). + * These values are intentionally wide because real-world setups + * often involve microphones, room acoustics, and cheap ADCs. + */ +#define SEP_TONE_MIN_HZ 1000 +#define SEP_TONE_MAX_HZ 3000 + +#define SEP_TOLERANCE_PULSES \ + (((SEP_TONE_MAX_HZ) - (SEP_TONE_MIN_HZ)) / (2 * (FRAME_RATE))) + +#define DATA_TONE_MIN_HZ 3000 +#define DATA_TONE_MAX_HZ 12000 + +/* Mid point used to distinguish the two data tones. */ +#define DATA_TONE_THRESHOLD_HZ 5000 + +/* + * Convert tone frequency ranges into pulse counts within the + * sliding analysis window. + * + * pulse_count = tone_frequency / FRAME_RATE + * where FRAME_RATE = SAMPLE_RATE / SAMPLES_PER_FRAME. + */ +#define FREQ_SEP_MIN ((SEP_TONE_MIN_HZ) / (FRAME_RATE)) +#define FREQ_SEP_MAX ((SEP_TONE_MAX_HZ) / (FRAME_RATE)) + +#define FREQ_DATA_MIN ((DATA_TONE_MIN_HZ) / (FRAME_RATE)) +#define FREQ_DATA_MAX ((DATA_TONE_MAX_HZ) / (FRAME_RATE)) + +#define FREQ_DATA_THRESHOLD ((DATA_TONE_THRESHOLD_HZ) / (FRAME_RATE)) + +/* + * These determine how long the program will wait during + * tone auto-detection, before shifting to defaults. + * It is done every LEARN_FRAMES number of frames. + */ +#define LEARN_SECONDS 1 +#define LEARN_FRAMES ((LEARN_SECONDS) * (FRAME_RATE)) + +/* + * Sample amplitude threshold used to convert the waveform + * into a pulse stream. Values near zero are regarded as noise. + */ +#define THRESHOLD 500 + +#define READ_BUF 4096 + +struct decoder_state { + unsigned char pulse[MAX_SAMPLES]; + + signed short inbuf[READ_BUF]; + size_t inpos; + size_t inlen; + + int ringpos; + int sep_pos; + + /* + * Sliding window pulse counters + * used to detect modem tones + */ + int freq_data; + int freq_separator; + int sample_count; + + int ascii_bit; + unsigned char ascii; + + int debug; + int swap_bytes; + + /* dynamic separator calibration */ + int sep_sum; + int sep_samples; + int sep_min; + int sep_max; + + /* for automatic tone detection */ + int freq_min; + int freq_max; + int freq_threshold; + int learn_frames; + + /* previous sample used for edge detection */ + signed short prev_sample; +}; + +static const char *argv0; + +/* + * 16-bit little endian words are read + * continuously. we will swap them, if + * the host cpu is big endian. + */ +static int host_is_big_endian(void); + +/* main loop */ +static void handle_audio(struct decoder_state *st); + +/* separate tone tolerances */ +static void select_separator_tone(struct decoder_state *st); +static int is_valid_signal(struct decoder_state *st); + +/* output to terminal */ +static int set_ascii_bit(struct decoder_state *st); +static void print_char(struct decoder_state *st); +static void reset_char(struct decoder_state *st); + +/* process samples/frames */ +static void decode_pulse(struct decoder_state *st); +static signed short read_sample(struct decoder_state *st); +static void read_words(struct decoder_state *st); + +/* continually adjust tone */ +static void detect_tone(struct decoder_state *st); +static int silent_signal(struct decoder_state *st); +static void select_low_tone(struct decoder_state *st); + +/* debug */ +static void print_stats(struct decoder_state *st); + +/* error handling / usage */ +static void err(int errval, const char *msg, ...); +static void usage(void); +static const char *progname(void); + +/* portability (old systems) */ +int getopt(int, char * const *, const char *); +extern char *optarg; +extern int optind; +extern int opterr; +extern int optopt; + +#ifndef CHAR_BIT +#define CHAR_BIT 8 +#endif + +typedef char static_assert_char_is_8_bits[(CHAR_BIT == 8) ? 1 : -1]; +typedef char static_assert_char_is_1[(sizeof(char) == 1) ? 1 : -1]; +typedef char static_assert_short[(sizeof(short) == 2) ? 1 : -1]; +typedef char static_assert_int_is_4[(sizeof(int) >= 4) ? 1 : -1]; +typedef char static_assert_twos_complement[ + ((-1 & 3) == 3) ? 1 : -1 +]; + +int +main(int argc, char **argv) +{ + struct decoder_state st; + int c; + + argv0 = argv[0]; + +#if defined (__OpenBSD__) && defined(OpenBSD) +#if OpenBSD >= 509 + if (pledge("stdio", NULL) == -1) + err(errno, "pledge"); +#endif +#endif + + memset(&st, 0, sizeof(st)); + + while ((c = getopt(argc, argv, "d")) != -1) { + if (c != 'd') + usage(); + st.debug = 1; + break; + } + + /* fallback in case tone detection fails */ + st.freq_min = 100000; + st.freq_max = 0; + st.freq_threshold = FREQ_DATA_THRESHOLD; + + /* + * Used for separator calibration + */ + st.sep_min = FREQ_SEP_MIN; + st.sep_max = FREQ_SEP_MAX; + + st.ascii_bit = 7; + + st.ringpos = 0; + st.sep_pos = SAMPLES_PER_FRAME; + + if (host_is_big_endian()) + st.swap_bytes = 1; + + setvbuf(stdout, NULL, _IONBF, 0); + + for (;;) + handle_audio(&st); + + return EXIT_SUCCESS; +} + +static int +host_is_big_endian(void) +{ + unsigned int x = 1; + return (*(unsigned char *)&x == 0); +} + +static void +handle_audio(struct decoder_state *st) +{ + int sample; + + /* + * If the modem signal disappears for several (read: 3) + * frames, discard the partially assembled character. + */ + if (st->sample_count >= (3 * SAMPLES_PER_FRAME) || + st->freq_separator <= 0) + reset_char(st); + + st->sample_count = 0; + + /* process exactly one frame */ + for (sample = 0; sample < SAMPLES_PER_FRAME; sample++) + decode_pulse(st); + + select_separator_tone(st); + + if (set_ascii_bit(st) < 0) + print_char(st); + + /* Detect tone per each frame */ + detect_tone(st); +} + +/* + * collect separator tone statistics + * (and auto-adjust tolerances) + */ +static void +select_separator_tone(struct decoder_state *st) +{ + int avg; + + if (!is_valid_signal(st)) + return; + + st->sep_sum += st->freq_separator; + st->sep_samples++; + + if (st->sep_samples != 50) + return; + + avg = st->sep_sum / st->sep_samples; + + st->sep_min = avg - SEP_TOLERANCE_PULSES; + st->sep_max = avg + SEP_TOLERANCE_PULSES; + + /* reset calibration accumulators */ + st->sep_sum = 0; + st->sep_samples = 0; + + if (st->debug) + printf("separator calibrated: %dHz\n", + avg * FRAME_RATE); +} + +/* + * Verify that the observed pulse densities fall within the + * expected ranges for spkmodem tones. This prevents random noise + * from being misinterpreted as data. + */ +static int +is_valid_signal(struct decoder_state *st) +{ + if (st->freq_data <= 0) + return 0; + + if (st->freq_separator < st->sep_min || + st->freq_separator > st->sep_max) + return 0; + + return 1; +} + +/* + * Each validated frame contributes one bit of modem data. + * Bits are accumulated MSB-first into the ASCII byte. + */ +static int +set_ascii_bit(struct decoder_state *st) +{ + if (st->debug) + print_stats(st); + + if (!is_valid_signal(st)) + return st->ascii_bit; + + if (st->freq_data < st->freq_threshold) + st->ascii |= (1 << st->ascii_bit); + + st->ascii_bit--; + + return st->ascii_bit; +} + +static void +print_char(struct decoder_state *st) +{ + if (st->debug) + printf("<%c,%x>", st->ascii, st->ascii); + else + putchar(st->ascii); + + reset_char(st); +} + +static void +reset_char(struct decoder_state *st) +{ + st->ascii = 0; + st->ascii_bit = 7; +} + +/* + * Main demodulation step (moving-sum FIR filter). + */ +static void +decode_pulse(struct decoder_state *st) +{ + unsigned char old_ring, old_sep; + unsigned char new_pulse; + signed short sample; + int ringpos; + int sep_pos; + int diff_edge; + int diff_amp; + + ringpos = st->ringpos; + sep_pos = st->sep_pos; + + /* + * Sliding rectangular FIR (Finite Impulse Response) filter. + * + * After thresholding, the signal becomes a stream of 0/1 pulses. + * The decoder measures pulse density over two windows: + * + * freq_data: pulses in the "data" window + * freq_separator: pulses in the "separator" window + * + * Instead of calculating each window every time (O(N) per frame), we + * update the window sums incrementally: + * + * sum_new = sum_old - pulse_leaving + pulse_entering + * + * This keeps the filter O(1) per sample instead of O(N). + * The technique is equivalent to a rectangular FIR filter + * implemented as a sliding moving sum. + * + * The two windows are exactly SAMPLES_PER_FRAME apart in the ring + * buffer, so the pulse leaving the data window is simultaneously + * entering the separator window. + */ + old_ring = st->pulse[ringpos]; + old_sep = st->pulse[sep_pos]; + st->freq_data -= old_ring; + st->freq_data += old_sep; + st->freq_separator -= old_sep; + + sample = read_sample(st); + + /* + * Avoid startup edge. Since + * it's zero at startup, this + * may wrongly produce a pulse + */ + if (st->sample_count == 0) + st->prev_sample = sample; + + /* + * Detect edges instead of amplitude. + * This is more tolerant of weak microphones + * and speaker distortion.. + * + * However, we check both slope edges and + * amplitude, to mitagate noise. + */ + diff_amp = sample; + diff_edge = sample - st->prev_sample; + if (diff_edge < 0) + diff_edge = -diff_edge; + if (diff_amp < 0) + diff_amp = -diff_amp; + if (diff_edge > THRESHOLD && + diff_amp > THRESHOLD) + new_pulse = 1; + else + new_pulse = 0; + st->prev_sample = sample; + + st->pulse[ringpos] = new_pulse; + st->freq_separator += new_pulse; + + /* + * Advance both FIR windows through the ring buffer. + * The separator window always stays one frame ahead + * of the data window. + */ + if (++ringpos >= MAX_SAMPLES) + ringpos = 0; + if (++sep_pos >= MAX_SAMPLES) + sep_pos = 0; + + st->ringpos = ringpos; + st->sep_pos = sep_pos; + + st->sample_count++; +} + +static signed short +read_sample(struct decoder_state *st) +{ + signed short sample; + unsigned short u; + + while (st->inpos >= st->inlen) + read_words(st); + + sample = st->inbuf[st->inpos++]; + + if (st->swap_bytes) { + u = (unsigned short)sample; + u = (u >> 8) | (u << 8); + + sample = (signed short)u; + } + + return sample; +} + +static void +read_words(struct decoder_state *st) +{ + size_t n; + + n = fread(st->inbuf, sizeof(st->inbuf[0]), + READ_BUF, stdin); + + if (n != 0) { + st->inpos = 0; + st->inlen = n; + + return; + } + + if (ferror(stdin)) + err(errno, "stdin read"); + if (feof(stdin)) + exit(EXIT_SUCCESS); +} + +/* + * Automatically detect spkmodem tone + */ +static void +detect_tone(struct decoder_state *st) +{ + if (st->learn_frames >= LEARN_FRAMES) + return; + + st->learn_frames++; + + if (silent_signal(st)) + return; + + select_low_tone(st); + + if (st->learn_frames != LEARN_FRAMES) + return; + + /* + * If the observed frequencies are too close, + * learning likely failed (only one tone seen). + * Keep the default threshold. + */ + if (st->freq_max - st->freq_min < 2) + return; + + st->freq_threshold = + (st->freq_min + st->freq_max) / 2; + + if (st->debug) + printf("auto threshold: %dHz\n", + st->freq_threshold * FRAME_RATE); +} + +/* + * Ignore silence / near silence. + * Both FIR windows will be near zero when no signal exists. + */ +static int +silent_signal(struct decoder_state *st) +{ + return (st->freq_data <= 2 && + st->freq_separator <= 2); +} + +/* + * Choose the lowest active tone. + * Separator frames carry tone in the separator window, + * data frames carry tone in the data window. + */ +static void +select_low_tone(struct decoder_state *st) +{ + int f; + + f = st->freq_data; + + if (f <= 0 || (st->freq_separator > 0 && + st->freq_separator < f)) + f = st->freq_separator; + + if (f <= 0) + return; + + if (f < st->freq_min) + st->freq_min = f; + + if (f > st->freq_max) + st->freq_max = f; +} + +static void +print_stats(struct decoder_state *st) +{ + long pos; + + int data_hz = st->freq_data * FRAME_RATE; + int sep_hz = st->freq_separator * FRAME_RATE; + int sep_hz_min = st->sep_min * FRAME_RATE; + int sep_hz_max = st->sep_max * FRAME_RATE; + + if ((pos = ftell(stdin)) == -1) { + printf("%d %d %d data=%dHz sep=%dHz(min %dHz %dHz)\n", + st->freq_data, + st->freq_separator, + st->freq_threshold, + data_hz, + sep_hz, + sep_hz_min, + sep_hz_max); + return; + } + + printf("%d %d %d @%ld data=%dHz sep=%dHz(min %dHz %dHz)\n", + st->freq_data, + st->freq_separator, + st->freq_threshold, + pos - SAMPLE_OFFSET, + data_hz, + sep_hz, + sep_hz_min, + sep_hz_max); +} + +static void +err(int errval, const char *msg, ...) +{ + va_list ap; + + fprintf(stderr, "%s: ", progname()); + + va_start(ap, msg); + vfprintf(stderr, msg, ap); + va_end(ap); + + fprintf(stderr, ": %s\n", strerror(errval)); + exit(EXIT_FAILURE); +} + +static void +usage(void) +{ + fprintf(stderr, "usage: %s [-d]\n", progname()); + exit(EXIT_FAILURE); +} + +static const char * +progname(void) +{ + const char *p; + + if (argv0 == NULL || *argv0 == '\0') + return ""; + + p = strrchr(argv0, '/'); + + if (p) + return p + 1; + else + return argv0; +} |