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>

3 files changed, 528 insertions, 0 deletions

diff --git a/util/spkmodem_decode/.gitignore b/util/spkmodem_decode/.gitignore
new file mode 100644
index 00000000..42814fe4
--- /dev/null
+++ b/util/spkmodem_decode/.gitignore
@@ -0,0 +1,2 @@
+spkmodem-recv
+spkmodem-decode
diff --git a/util/spkmodem_decode/Makefile b/util/spkmodem_decode/Makefile
new file mode 100644
index 00000000..b00c4f43
--- /dev/null
+++ b/util/spkmodem_decode/Makefile
@@ -0,0 +1,30 @@
+# SPDX-License-Identifier: MIT
+# SPDX-FileCopyrightText: 2022,2026 Leah Rowe <leah@libreboot.org>
+# SPDX-FileCopyrightText: 2023 Riku Viitanen <riku.viitanen@protonmail.com>
+
+CC?=cc
+CFLAGS?=-Os -Wall -Wextra -Werror -pedantic -std=c90
+DESTDIR?=
+PREFIX?=/usr/local
+INSTALL?=install
+
+PROG=spkmodem-decode
+
+all: $(PROG)
+
+$(PROG): spkmodem-decode.c
+	$(CC) $(CFLAGS) spkmodem-decode.c -o $(PROG)
+
+install: $(PROG)
+	mkdir -p $(DESTDIR)$(PREFIX)/bin/
+	install -c $(PROG) $(DESTDIR)$(PREFIX)/bin/
+
+uninstall:
+	rm -f $(DESTDIR)$(PREFIX)/bin/$(PROG)
+
+clean:
+	rm -f $(PROG)
+
+distclean: clean
+
+.PHONY: all install uninstall clean distclean
diff --git a/util/spkmodem_decode/spkmodem-decode.c b/util/spkmodem_decode/spkmodem-decode.c
new file mode 100644
index 00000000..820fa61e
--- /dev/null
+++ b/util/spkmodem_decode/spkmodem-decode.c
@@ -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 ≈ 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;
+}