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refactoring

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This commit is contained in:
Maxim Devaev
2025-01-17 20:40:18 +02:00
parent 29c98e3908
commit ba246d90c0
7 changed files with 165 additions and 165 deletions
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92
janus/src/plugin.c
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@@ -48,7 +48,7 @@
#include "const.h"
#include "logging.h"
#include "client.h"
#include "audio.h"
#include "acap.h"
#include "rtp.h"
#include "rtpv.h"
#include "rtpa.h"
@@ -69,11 +69,11 @@ static pthread_t _g_video_rtp_tid;
static atomic_bool _g_video_rtp_tid_created = false;
static pthread_t _g_video_sink_tid;
static atomic_bool _g_video_sink_tid_created = false;
static pthread_t _g_audio_tid;
static atomic_bool _g_audio_tid_created = false;
static pthread_t _g_acap_tid;
static atomic_bool _g_acap_tid_created = false;
static pthread_mutex_t _g_video_lock = PTHREAD_MUTEX_INITIALIZER;
static pthread_mutex_t _g_audio_lock = PTHREAD_MUTEX_INITIALIZER;
static pthread_mutex_t _g_acap_lock = PTHREAD_MUTEX_INITIALIZER;
static atomic_bool _g_ready = false;
static atomic_bool _g_stop = false;
static atomic_bool _g_has_watchers = false;
@@ -84,11 +84,11 @@ static atomic_bool _g_key_required = false;
#define _LOCK_VIDEO US_MUTEX_LOCK(_g_video_lock)
#define _UNLOCK_VIDEO US_MUTEX_UNLOCK(_g_video_lock)
#define _LOCK_AUDIO US_MUTEX_LOCK(_g_audio_lock)
#define _UNLOCK_AUDIO US_MUTEX_UNLOCK(_g_audio_lock)
#define _LOCK_ACAP US_MUTEX_LOCK(_g_acap_lock)
#define _UNLOCK_ACAP US_MUTEX_UNLOCK(_g_acap_lock)
#define _LOCK_ALL { _LOCK_VIDEO; _LOCK_AUDIO; }
#define _UNLOCK_ALL { _UNLOCK_AUDIO; _UNLOCK_VIDEO; }
#define _LOCK_ALL { _LOCK_VIDEO; _LOCK_ACAP; }
#define _UNLOCK_ALL { _UNLOCK_ACAP; _UNLOCK_VIDEO; }
#define _READY atomic_load(&_g_ready)
#define _STOP atomic_load(&_g_stop)
@@ -198,15 +198,15 @@ static void *_video_sink_thread(void *arg) {
return NULL;
}
static int _check_tc358743_audio(uint *audio_hz) {
static int _check_tc358743_acap(uint *hz) {
int fd;
if ((fd = open(_g_config->tc358743_dev_path, O_RDWR)) < 0) {
US_JLOG_PERROR("audio", "Can't open TC358743 V4L2 device");
US_JLOG_PERROR("acap", "Can't open TC358743 V4L2 device");
return -1;
}
const int checked = us_tc358743_xioctl_get_audio_hz(fd, audio_hz);
const int checked = us_tc358743_xioctl_get_audio_hz(fd, hz);
if (checked < 0) {
US_JLOG_PERROR("audio", "Can't check TC358743 audio state (%d)", checked);
US_JLOG_PERROR("acap", "Can't check TC358743 audio state (%d)", checked);
close(fd);
return -1;
}
@@ -214,12 +214,12 @@ static int _check_tc358743_audio(uint *audio_hz) {
return 0;
}
static void *_audio_thread(void *arg) {
static void *_acap_thread(void *arg) {
(void)arg;
US_THREAD_SETTLE("us_audio");
atomic_store(&_g_audio_tid_created, true);
US_THREAD_SETTLE("us_acap");
atomic_store(&_g_acap_tid_created, true);
assert(_g_config->audio_dev_name != NULL);
assert(_g_config->acap_dev_name != NULL);
assert(_g_config->tc358743_dev_path != NULL);
int once = 0;
@@ -230,42 +230,42 @@ static void *_audio_thread(void *arg) {
continue;
}
uint audio_hz = 0;
us_audio_s *audio = NULL;
uint hz = 0;
us_acap_s *acap = NULL;
if (_check_tc358743_audio(&audio_hz) < 0) {
goto close_audio;
if (_check_tc358743_acap(&hz) < 0) {
goto close_acap;
}
if (audio_hz == 0) {
US_ONCE({ US_JLOG_INFO("audio", "No audio presented from the host"); });
goto close_audio;
if (hz == 0) {
US_ONCE({ US_JLOG_INFO("acap", "No audio presented from the host"); });
goto close_acap;
}
US_ONCE({ US_JLOG_INFO("audio", "Detected host audio"); });
if ((audio = us_audio_init(_g_config->audio_dev_name, audio_hz)) == NULL) {
goto close_audio;
US_ONCE({ US_JLOG_INFO("acap", "Detected host audio"); });
if ((acap = us_acap_init(_g_config->acap_dev_name, hz)) == NULL) {
goto close_acap;
}
once = 0;
while (!_STOP && _HAS_WATCHERS && _HAS_LISTENERS) {
if (_check_tc358743_audio(&audio_hz) < 0 || audio->pcm_hz != audio_hz) {
goto close_audio;
if (_check_tc358743_acap(&hz) < 0 || acap->pcm_hz != hz) {
goto close_acap;
}
uz size = US_RTP_DATAGRAM_SIZE - US_RTP_HEADER_SIZE;
u8 data[size];
u64 pts;
const int result = us_audio_get_encoded(audio, data, &size, &pts);
const int result = us_acap_get_encoded(acap, data, &size, &pts);
if (result == 0) {
_LOCK_AUDIO;
_LOCK_ACAP;
us_rtpa_wrap(_g_rtpa, data, size, pts);
_UNLOCK_AUDIO;
_UNLOCK_ACAP;
} else if (result == -1) {
goto close_audio;
goto close_acap;
}
}
close_audio:
US_DELETE(audio, us_audio_destroy);
close_acap:
US_DELETE(acap, us_acap_destroy);
sleep(1); // error_delay
}
return NULL;
@@ -292,9 +292,9 @@ static int _plugin_init(janus_callbacks *gw, const char *config_dir_path) {
US_RING_INIT_WITH_ITEMS(_g_video_ring, 64, us_frame_init);
_g_rtpv = us_rtpv_init(_relay_rtp_clients);
if (_g_config->audio_dev_name != NULL && us_audio_probe(_g_config->audio_dev_name)) {
if (_g_config->acap_dev_name != NULL && us_acap_probe(_g_config->acap_dev_name)) {
_g_rtpa = us_rtpa_init(_relay_rtp_clients);
US_THREAD_CREATE(_g_audio_tid, _audio_thread, NULL);
US_THREAD_CREATE(_g_acap_tid, _acap_thread, NULL);
}
US_THREAD_CREATE(_g_video_rtp_tid, _video_rtp_thread, NULL);
US_THREAD_CREATE(_g_video_sink_tid, _video_sink_thread, NULL);
@@ -310,7 +310,7 @@ static void _plugin_destroy(void) {
# define JOIN(_tid) { if (atomic_load(&_tid##_created)) { US_THREAD_JOIN(_tid); } }
JOIN(_g_video_sink_tid);
JOIN(_g_video_rtp_tid);
JOIN(_g_audio_tid);
JOIN(_g_acap_tid);
# undef JOIN
US_LIST_ITERATE(_g_clients, client, {
@@ -351,7 +351,7 @@ static void _plugin_destroy_session(janus_plugin_session* session, int *err) {
found = true;
} else {
has_watchers = (has_watchers || atomic_load(&client->transmit));
has_listeners = (has_listeners || atomic_load(&client->transmit_audio));
has_listeners = (has_listeners || atomic_load(&client->transmit_acap));
}
});
if (!found) {
@@ -459,21 +459,21 @@ static struct janus_plugin_result *_plugin_handle_message(
} else if (!strcmp(request_str, "watch")) {
uint video_orient = 0;
bool with_audio = false;
bool with_mic = false;
bool with_acap = false;
bool with_aplay = false;
{
json_t *const params = json_object_get(msg, "params");
if (params != NULL) {
{
json_t *const obj = json_object_get(params, "audio");
if (obj != NULL && json_is_boolean(obj)) {
with_audio = (_g_rtpa != NULL && json_boolean_value(obj));
with_acap = (_g_rtpa != NULL && json_boolean_value(obj));
}
}
{
json_t *const obj = json_object_get(params, "microphone");
if (obj != NULL && json_is_boolean(obj)) {
with_mic = (with_audio && json_boolean_value(obj)); // FIXME: also check playback
with_aplay = (with_acap && json_boolean_value(obj)); // FIXME: also check playback
}
}
{
@@ -492,7 +492,7 @@ static struct janus_plugin_result *_plugin_handle_message(
{
char *sdp;
char *const video_sdp = us_rtpv_make_sdp(_g_rtpv);
char *const audio_sdp = (with_audio ? us_rtpa_make_sdp(_g_rtpa, with_mic) : us_strdup(""));
char *const audio_sdp = (with_acap ? us_rtpa_make_sdp(_g_rtpa, with_aplay) : us_strdup(""));
US_ASPRINTF(sdp,
"v=0" RN
"o=- %" PRIu64 " 1 IN IP4 0.0.0.0" RN
@@ -523,17 +523,17 @@ static struct janus_plugin_result *_plugin_handle_message(
bool has_listeners = false;
US_LIST_ITERATE(_g_clients, client, {
if (client->session == session) {
atomic_store(&client->transmit_audio, with_audio);
atomic_store(&client->transmit_acap, with_acap);
atomic_store(&client->video_orient, video_orient);
}
has_listeners = (has_listeners || atomic_load(&client->transmit_audio));
has_listeners = (has_listeners || atomic_load(&client->transmit_acap));
});
atomic_store(&_g_has_listeners, has_listeners);
_UNLOCK_ALL;
}
} else if (!strcmp(request_str, "features")) {
json_t *const features = json_pack("{sb}", "audio", (_g_rtpa != NULL));
json_t *const features = json_pack("{sbsb}", "audio", (_g_rtpa != NULL), "microphone", false);
PUSH_STATUS("features", features, NULL);
json_decref(features);