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/*
* Copyright (C) 2020 Mark Hills <mark@xwax.org>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* version 2, as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License version 2 for more details.
*
* You should have received a copy of the GNU General Public License
* version 2 along with this program; if not, write to the Free
* Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
* MA 02110-1301, USA.
*
*/
#include <netdb.h>
#include <string.h>
#include <alsa/asoundlib.h>
#include <opus/opus.h>
#include <ortp/ortp.h>
#include <sys/socket.h>
#include <sys/types.h>
#include "defaults.h"
#include "device.h"
#include "notice.h"
#include "sched.h"
static unsigned int verbose = DEFAULT_VERBOSE;
static RtpSession* create_rtp_send(const char *addr_desc, const int port)
{
RtpSession *session;
session = rtp_session_new(RTP_SESSION_SENDONLY);
assert(session != NULL);
rtp_session_set_scheduling_mode(session, 0);
rtp_session_set_blocking_mode(session, 0);
rtp_session_set_connected_mode(session, FALSE);
if (rtp_session_set_remote_addr(session, addr_desc, port) != 0)
abort();
if (rtp_session_set_payload_type(session, 0) != 0)
abort();
if (rtp_session_set_multicast_ttl(session, 16) != 0)
abort();
if (rtp_session_set_dscp(session, 40) != 0)
abort();
return session;
}
static int send_one_frame(snd_pcm_t *snd,
const unsigned int channels,
const snd_pcm_uframes_t samples,
OpusEncoder *encoder,
const size_t bytes_per_frame,
const unsigned int ts_per_frame,
RtpSession *session)
{
int16_t *pcm;
void *packet;
ssize_t z;
snd_pcm_sframes_t f;
static unsigned int ts = 0;
pcm = alloca(sizeof(*pcm) * samples * channels);
packet = alloca(bytes_per_frame);
f = snd_pcm_readi(snd, pcm, samples);
if (f < 0) {
if (f == -ESTRPIPE)
ts = 0;
f = snd_pcm_recover(snd, f, 0);
if (f < 0) {
aerror("snd_pcm_readi", f);
return -1;
}
return 0;
}
/* Opus encoder requires a complete frame, so if we xrun
* mid-frame then we discard the incomplete audio. The next
* read will catch the error condition and recover */
if (f < samples) {
fprintf(stderr, "Short read, %ld\n", f);
return 0;
}
z = opus_encode(encoder, pcm, samples, packet, bytes_per_frame);
if (z < 0) {
fprintf(stderr, "opus_encode_float: %s\n", opus_strerror(z));
return -1;
}
rtp_session_send_with_ts(session, packet, z, ts);
ts += ts_per_frame;
return 0;
}
static int run_tx(snd_pcm_t *snd,
const unsigned int channels,
const snd_pcm_uframes_t frame,
OpusEncoder *encoder,
const size_t bytes_per_frame,
const unsigned int ts_per_frame,
RtpSession *session)
{
for (;;) {
int r;
r = send_one_frame(snd, channels, frame,
encoder, bytes_per_frame, ts_per_frame,
session);
if (r == -1)
return -1;
if (verbose > 1)
fputc('>', stderr);
}
}
static void usage(FILE *fd)
{
fprintf(fd, "Usage: tx [<parameters>]\n"
"Real-time audio transmitter over IP\n");
fprintf(fd, "\nAudio device (ALSA) parameters:\n");
fprintf(fd, " -d <dev> Device name (default '%s')\n",
DEFAULT_DEVICE);
fprintf(fd, " -m <ms> Buffer time (default %d milliseconds)\n",
DEFAULT_BUFFER);
fprintf(fd, "\nNetwork parameters:\n");
fprintf(fd, " -h <addr> IP address to send to (default %s)\n",
DEFAULT_ADDR);
fprintf(fd, " -p <port> UDP port number (default %d)\n",
DEFAULT_PORT);
fprintf(fd, "\nEncoding parameters:\n");
fprintf(fd, " -r <rate> Sample rate (default %dHz)\n",
DEFAULT_RATE);
fprintf(fd, " -c <n> Number of channels (default %d)\n",
DEFAULT_CHANNELS);
fprintf(fd, " -f <n> Frame size (default %d samples, see below)\n",
DEFAULT_FRAME);
fprintf(fd, " -b <kbps> Bitrate (approx., default %d)\n",
DEFAULT_BITRATE);
fprintf(fd, "\nProgram parameters:\n");
fprintf(fd, " -v <n> Verbosity level (default %d)\n",
DEFAULT_VERBOSE);
fprintf(fd, " -D <file> Run as a daemon, writing process ID to the given file\n");
fprintf(fd, "\nAllowed frame sizes (-f) are defined by the Opus codec. For example,\n"
"at 48000Hz the permitted values are 120, 240, 480 or 960.\n");
}
int main(int argc, char *argv[])
{
int r, error;
size_t bytes_per_frame;
unsigned int ts_per_frame;
snd_pcm_t *snd;
OpusEncoder *encoder;
RtpSession *session;
/* command-line options */
const char *device = DEFAULT_DEVICE,
*addr = DEFAULT_ADDR,
*pid = NULL;
unsigned int buffer = DEFAULT_BUFFER,
rate = DEFAULT_RATE,
channels = DEFAULT_CHANNELS,
frame = DEFAULT_FRAME,
kbps = DEFAULT_BITRATE,
port = DEFAULT_PORT;
fputs(COPYRIGHT "\n", stderr);
for (;;) {
int c;
c = getopt(argc, argv, "b:c:d:f:h:m:p:r:v:D:");
if (c == -1)
break;
switch (c) {
case 'b':
kbps = atoi(optarg);
break;
case 'c':
channels = atoi(optarg);
break;
case 'd':
device = optarg;
break;
case 'f':
frame = atol(optarg);
break;
case 'h':
addr = optarg;
break;
case 'm':
buffer = atoi(optarg);
break;
case 'p':
port = atoi(optarg);
break;
case 'r':
rate = atoi(optarg);
break;
case 'v':
verbose = atoi(optarg);
break;
case 'D':
pid = optarg;
break;
default:
usage(stderr);
return -1;
}
}
encoder = opus_encoder_create(rate, channels, OPUS_APPLICATION_AUDIO,
&error);
if (encoder == NULL) {
fprintf(stderr, "opus_encoder_create: %s\n",
opus_strerror(error));
return -1;
}
bytes_per_frame = kbps * 1024 * frame / rate / 8;
/* Follow the RFC, payload 0 has 8kHz reference rate */
ts_per_frame = frame * 8000 / rate;
ortp_init();
ortp_scheduler_init();
session = create_rtp_send(addr, port);
assert(session != NULL);
r = snd_pcm_open(&snd, device, SND_PCM_STREAM_CAPTURE, 0);
if (r < 0) {
aerror("snd_pcm_open", r);
return -1;
}
if (set_alsa_hw(snd, rate, channels, buffer * 1000) == -1)
return -1;
if (set_alsa_sw(snd) == -1)
return -1;
if (pid)
go_daemon(pid);
go_realtime();
r = run_tx(snd, channels, frame, encoder, bytes_per_frame,
ts_per_frame, session);
if (snd_pcm_close(snd) < 0)
abort();
rtp_session_destroy(session);
ortp_exit();
ortp_global_stats_display();
opus_encoder_destroy(encoder);
return r;
}