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janus: using ring buffers for rtp pipelines

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This commit is contained in:
Maxim Devaev 2024-02-29 06:57:16 +02:00
parent 28cd5e5b87
commit 12937b93d5
6 changed files with 64 additions and 65 deletions
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95
janus/src/client.c
View File

@ -33,7 +33,7 @@
#include "uslibs/tools.h"
#include "uslibs/threading.h"
#include "uslibs/list.h"
#include "uslibs/queue.h"
#include "uslibs/ring.h"
#include "logging.h"
#include "rtp.h"
@ -54,10 +54,10 @@ us_janus_client_s *us_janus_client_init(janus_callbacks *gw, janus_plugin_sessio
atomic_init(&client->stop, false);
client->video_queue = us_queue_init(2048);
US_RING_INIT_WITH_ITEMS(client->video_ring, 2048, us_rtp_init);
US_THREAD_CREATE(client->video_tid, _video_thread, client);
client->audio_queue = us_queue_init(64);
US_RING_INIT_WITH_ITEMS(client->audio_ring, 64, us_rtp_init);
US_THREAD_CREATE(client->audio_tid, _audio_thread, client);
return client;
@ -65,14 +65,12 @@ us_janus_client_s *us_janus_client_init(janus_callbacks *gw, janus_plugin_sessio
void us_janus_client_destroy(us_janus_client_s *client) {
atomic_store(&client->stop, true);
us_queue_put(client->video_queue, NULL, 0);
us_queue_put(client->audio_queue, NULL, 0);
US_THREAD_JOIN(client->video_tid);
US_QUEUE_DELETE_WITH_ITEMS(client->video_queue, us_rtp_destroy);
US_RING_DELETE_WITH_ITEMS(client->video_ring, us_rtp_destroy);
US_THREAD_JOIN(client->audio_tid);
US_QUEUE_DELETE_WITH_ITEMS(client->audio_queue, us_rtp_destroy);
US_RING_DELETE_WITH_ITEMS(client->audio_ring, us_rtp_destroy);
free(client);
}
@ -82,12 +80,15 @@ void us_janus_client_send(us_janus_client_s *client, const us_rtp_s *rtp) {
atomic_load(&client->transmit)
&& (rtp->video || atomic_load(&client->transmit_audio))
) {
us_rtp_s *const new = us_rtp_dup(rtp);
if (us_queue_put((new->video ? client->video_queue : client->audio_queue), new, 0) != 0) {
US_JLOG_ERROR("client", "Session %p %s queue is full",
client->session, (new->video ? "video" : "audio"));
us_rtp_destroy(new);
us_ring_s *const ring = (rtp->video ? client->video_ring : client->audio_ring);
const int ri = us_ring_producer_acquire(ring, 0);
if (ri < 0) {
US_JLOG_ERROR("client", "Session %p %s ring is full",
client->session, (rtp->video ? "video" : "audio"));
return;
}
memcpy(ring->items[ri], rtp, sizeof(us_rtp_s));
us_ring_producer_release(ring, ri);
}
}
@ -101,45 +102,45 @@ static void *_audio_thread(void *v_client) {
static void *_common_thread(void *v_client, bool video) {
us_janus_client_s *const client = (us_janus_client_s *)v_client;
us_queue_s *const queue = (video ? client->video_queue : client->audio_queue);
assert(queue != NULL); // Audio may be NULL
us_ring_s *const ring = (video ? client->video_ring : client->audio_ring);
assert(ring != NULL); // Audio may be NULL
while (!atomic_load(&client->stop)) {
us_rtp_s *rtp;
if (!us_queue_get(queue, (void **)&rtp, 0.1)) {
if (rtp == NULL) {
break;
}
const int ri = us_ring_consumer_acquire(ring, 0.1);
if (ri < 0) {
continue;
}
us_rtp_s rtp;
memcpy(&rtp, ring->items[ri], sizeof(us_rtp_s));
us_ring_consumer_release(ring, ri);
if (
atomic_load(&client->transmit)
&& (video || atomic_load(&client->transmit_audio))
) {
janus_plugin_rtp packet = {
.video = rtp->video,
.buffer = (char *)rtp->datagram,
.length = rtp->used,
# if JANUS_PLUGIN_API_VERSION >= 100
// The uStreamer Janus plugin places video in stream index 0 and audio
// (if available) in stream index 1.
.mindex = (rtp->video ? 0 : 1),
# endif
};
janus_plugin_rtp_extensions_reset(&packet.extensions);
/*if (rtp->zero_playout_delay) {
// https://github.com/pikvm/pikvm/issues/784
packet.extensions.min_delay = 0;
packet.extensions.max_delay = 0;
} else {
packet.extensions.min_delay = 0;
// 10s - Chromium/WebRTC default
// 3s - Firefox default
packet.extensions.max_delay = 300; // == 3s, i.e. 10ms granularity
}*/
if (
atomic_load(&client->transmit)
&& (video || atomic_load(&client->transmit_audio))
) {
janus_plugin_rtp packet = {
.video = rtp.video,
.buffer = (char*)rtp.datagram,
.length = rtp.used,
# if JANUS_PLUGIN_API_VERSION >= 100
// The uStreamer Janus plugin places video in stream index 0 and audio
// (if available) in stream index 1.
.mindex = (rtp.video ? 0 : 1),
# endif
};
janus_plugin_rtp_extensions_reset(&packet.extensions);
/*if (rtp->zero_playout_delay) {
// https://github.com/pikvm/pikvm/issues/784
packet.extensions.min_delay = 0;
packet.extensions.max_delay = 0;
} else {
packet.extensions.min_delay = 0;
// 10s - Chromium/WebRTC default
// 3s - Firefox default
packet.extensions.max_delay = 300; // == 3s, i.e. 10ms granularity
}*/
client->gw->relay_rtp(client->session, &packet);
}
us_rtp_destroy(rtp);
client->gw->relay_rtp(client->session, &packet);
}
}
return NULL;

6
janus/src/client.h
View File

@ -29,7 +29,7 @@
#include "uslibs/types.h"
#include "uslibs/list.h"
#include "uslibs/queue.h"
#include "uslibs/ring.h"
#include "rtp.h"
@ -44,8 +44,8 @@ typedef struct us_janus_client_sx {
pthread_t audio_tid;
atomic_bool stop;
us_queue_s *video_queue;
us_queue_s *audio_queue;
us_ring_s *video_ring;
us_ring_s *audio_ring;
US_LIST_STRUCT(struct us_janus_client_sx);
} us_janus_client_s;

18
janus/src/rtp.c
View File

@ -31,26 +31,22 @@
#include "uslibs/tools.h"
us_rtp_s *us_rtp_init(uint payload, bool video) {
us_rtp_s *us_rtp_init(void) {
us_rtp_s *rtp;
US_CALLOC(rtp, 1);
rtp->payload = payload;
rtp->video = video;
rtp->ssrc = us_triple_u32(us_get_now_monotonic_u64());
return rtp;
}
us_rtp_s *us_rtp_dup(const us_rtp_s *rtp) {
us_rtp_s *new;
US_CALLOC(new, 1);
memcpy(new, rtp, sizeof(us_rtp_s));
return new;
}
void us_rtp_destroy(us_rtp_s *rtp) {
free(rtp);
}
void us_rtp_assign(us_rtp_s *rtp, uint payload, bool video) {
rtp->payload = payload;
rtp->video = video;
rtp->ssrc = us_triple_u32(us_get_now_monotonic_u64());
}
void us_rtp_write_header(us_rtp_s *rtp, u32 pts, bool marked) {
u32 word0 = 0x80000000;
if (marked) {

4
janus/src/rtp.h
View File

@ -44,8 +44,8 @@ typedef struct {
typedef void (*us_rtp_callback_f)(const us_rtp_s *rtp);
us_rtp_s *us_rtp_init(uint payload, bool video);
us_rtp_s *us_rtp_dup(const us_rtp_s *rtp);
us_rtp_s *us_rtp_init(void);
void us_rtp_destroy(us_rtp_s *rtp);
void us_rtp_assign(us_rtp_s *rtp, uint payload, bool video);
void us_rtp_write_header(us_rtp_s *rtp, u32 pts, bool marked);

3
janus/src/rtpa.c
View File

@ -32,7 +32,8 @@
us_rtpa_s *us_rtpa_init(us_rtp_callback_f callback) {
us_rtpa_s *rtpa;
US_CALLOC(rtpa, 1);
rtpa->rtp = us_rtp_init(111, false);
rtpa->rtp = us_rtp_init();
us_rtp_assign(rtpa->rtp, 111, false);
rtpa->callback = callback;
return rtpa;
}

3
janus/src/rtpv.c
View File

@ -44,7 +44,8 @@ static sz _find_annexb(const u8 *data, uz size);
us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback) {
us_rtpv_s *rtpv;
US_CALLOC(rtpv, 1);
rtpv->rtp = us_rtp_init(96, true);
rtpv->rtp = us_rtp_init();
us_rtp_assign(rtpv->rtp, 96, true);
rtpv->callback = callback;
return rtpv;
}