rename util/spkmodem-recv to spkmodem-decode

it's no longer resembling the original util at all,
so a rename seems indicated. yes.

Signed-off-by: Leah Rowe <leah@libreboot.org>
This commit is contained in:
Leah Rowe
2026-03-12 21:21:36 +00:00
parent a31cfa4592
commit 30d5ca5146
4 changed files with 16 additions and 6 deletions
+496
View File
@@ -0,0 +1,496 @@
/*
* 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 <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
*/
#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 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 is: 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))
/*
* 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;
};
static const char *argv0;
static int host_is_big_endian(void);
static void handle_audio(struct decoder_state *st);
static int valid_signal(struct decoder_state *st);
static void decode_pulse(struct decoder_state *st);
static signed short read_sample(struct decoder_state *st);
static int set_ascii_bit(struct decoder_state *st);
static void print_char(struct decoder_state *st);
static void print_stats(struct decoder_state *st);
static void reset_char(struct decoder_state *st);
static void err(int errval, const char *msg, ...);
static void usage(void);
static const char *progname(void);
int getopt(int, char * const *, const char *);
extern char *optarg;
extern int optind;
extern int opterr;
extern int optopt;
int
main(int argc, char **argv)
{
struct decoder_state st;
int c;
#if defined (__OpenBSD__) && defined(OpenBSD)
#if OpenBSD >= 509
if (pledge("stdio", NULL) == -1)
err(errno, "pledge");
#endif
#endif
memset(&st, 0, sizeof(st));
st.ascii_bit = 7;
st.ringpos = 0;
st.sep_pos = SAMPLES_PER_FRAME;
argv0 = argv[0];
while ((c = getopt(argc, argv, "d")) != -1) {
if (c != 'd')
usage();
st.debug = 1;
break;
}
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 frames,
* discard the partially assembled character.
*/
if (st->sample_count > (3 * SAMPLES_PER_FRAME))
reset_char(st);
if (!valid_signal(st)) {
decode_pulse(st);
return;
}
if (set_ascii_bit(st) < 0)
print_char(st);
st->sample_count = 0;
for (sample = 0; sample < SAMPLES_PER_FRAME; sample++)
decode_pulse(st);
}
/*
* 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
valid_signal(struct decoder_state *st)
{
return (st->freq_separator > FREQ_SEP_MIN &&
st->freq_separator < FREQ_SEP_MAX &&
st->freq_data > FREQ_DATA_MIN &&
st->freq_data < FREQ_DATA_MAX);
}
/*
* 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;
int ringpos;
int sep_pos;
signed short sample;
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);
/*
* Convert the waveform sample into a pulse (0 or 1).
*
* The unsigned comparison creates a small dead zone near zero,
* suppressing small amplitude noise from microphones or
* cheap ADCs. Real PC speaker tones are far outside this
* range, so they still produce clean pulses.
*/
if ((unsigned)(sample + THRESHOLD)
> (unsigned)(2 * THRESHOLD))
new_pulse = 1;
else
new_pulse = 0;
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.
*/
ringpos++;
if (ringpos >= MAX_SAMPLES)
ringpos = 0;
sep_pos++;
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)
{
size_t n;
signed short sample;
unsigned short u;
while (st->inpos >= st->inlen) {
n = fread(st->inbuf, sizeof(st->inbuf[0]),
READ_BUF, stdin);
if (n == 0) {
if (ferror(stdin))
err(errno, "stdin read");
if (feof(stdin))
exit(EXIT_SUCCESS);
}
st->inpos = 0;
st->inlen = n;
}
sample = st->inbuf[st->inpos++];
if (st->swap_bytes) {
u = (unsigned short)sample;
u = (u >> 8) | (u << 8);
sample = (signed short)u;
}
return sample;
}
/*
* 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 (st->freq_data < FREQ_DATA_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
print_stats(struct decoder_state *st)
{
long pos;
if ((pos = ftell(stdin)) == -1) {
printf("%d %d %d\n",
st->freq_data,
st->freq_separator,
FREQ_DATA_THRESHOLD);
return;
}
printf("%d %d %d @%ld\n",
st->freq_data,
st->freq_separator,
FREQ_DATA_THRESHOLD,
pos - SAMPLE_OFFSET);
}
static void
reset_char(struct decoder_state *st)
{
st->ascii = 0;
st->ascii_bit = 7;
}
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;
}