forked/ustreamer
1
0
Fork 0
mirror of https://github.com/pikvm/ustreamer.git synced 2026-02-18 02:55:46 +00:00
Code Issues Packages Projects Releases Wiki Activity

auto playout delay

Browse Source
This commit is contained in:
Maxim Devaev 2022-11-13 06:26:30 +03:00
parent c1363d55e0
commit 414f536ace
9 changed files with 19 additions and 22 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

7
janus/src/client.c
View File

@ -108,12 +108,13 @@ static void *_common_thread(void *v_client, bool video) {
packet.mindex = (rtp->video ? 0 : 1);
# endif
janus_plugin_rtp_extensions_reset(&packet.extensions);
// FIXME: Это очень эффективный способ уменьшить задержку, но WebRTC стек в хроме и фоксе
// слишком корявый, чтобы обработать это, из-за чего на кейфреймах начинаются заикания.
// - https://github.com/Glimesh/janus-ftl-plugin/issues/101
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;
packet.extensions.max_delay = 1000;
}
client->gw->relay_rtp(client->session, &packet);
}

9
janus/src/config.c
View File

@ -24,7 +24,7 @@
static char *_get_value(janus_config *jcfg, const char *section, const char *option);
static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def);
// static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def);
us_config_s *us_config_init(const char *config_dir_path) {
@ -51,9 +51,6 @@ us_config_s *us_config_init(const char *config_dir_path) {
US_JLOG_ERROR("config", "Missing config value: video.sink (ex. memsink.object)");
goto error;
}
if ((config->video_zero_playout_delay = _get_bool(jcfg, "video", "zero_playout_delay", false)) == true) {
US_JLOG_INFO("config", "Enabled the experimental Playout-Delay=0 RTP extension for VIDEO");
}
if ((config->audio_dev_name = _get_value(jcfg, "audio", "device")) != NULL) {
US_JLOG_INFO("config", "Enabled the experimental AUDIO feature");
if ((config->tc358743_dev_path = _get_value(jcfg, "audio", "tc358743")) == NULL) {
@ -88,7 +85,7 @@ static char *_get_value(janus_config *jcfg, const char *section, const char *opt
return us_strdup(option_obj->value);
}
static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def) {
/*static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def) {
char *const tmp = _get_value(jcfg, section, option);
bool value = def;
if (tmp != NULL) {
@ -96,4 +93,4 @@ static bool _get_bool(janus_config *jcfg, const char *section, const char *optio
free(tmp);
}
return value;
}
}*/

1
janus/src/config.h
View File

@ -36,7 +36,6 @@
typedef struct {
char *video_sink_name;
bool video_zero_playout_delay;
char *audio_dev_name;
char *tc358743_dev_path;

2
janus/src/plugin.c
View File

@ -262,7 +262,7 @@ static int _plugin_init(janus_callbacks *gw, const char *config_dir_path) {
_g_gw = gw;
_g_video_queue = us_queue_init(1024);
_g_rtpv = us_rtpv_init(_relay_rtp_clients, _g_config->video_zero_playout_delay);
_g_rtpv = us_rtpv_init(_relay_rtp_clients);
if (_g_config->audio_dev_name != NULL) {
_g_rtpa = us_rtpa_init(_relay_rtp_clients);
US_THREAD_CREATE(_g_audio_tid, _audio_thread, NULL);

3
janus/src/rtp.c
View File

@ -26,12 +26,11 @@
#include "rtp.h"
us_rtp_s *us_rtp_init(unsigned payload, bool video, bool zero_playout_delay) {
us_rtp_s *us_rtp_init(unsigned payload, bool video) {
us_rtp_s *rtp;
US_CALLOC(rtp, 1);
rtp->payload = payload;
rtp->video = video;
rtp->zero_playout_delay = zero_playout_delay; // See client.c
rtp->ssrc = us_triple_u32(us_get_now_monotonic_u64());
return rtp;
}

4
janus/src/rtp.h
View File

@ -39,18 +39,18 @@
typedef struct {
unsigned payload;
bool video;
bool zero_playout_delay;
uint32_t ssrc;
uint16_t seq;
uint8_t datagram[US_RTP_DATAGRAM_SIZE];
size_t used;
bool zero_playout_delay;
} us_rtp_s;
typedef void (*us_rtp_callback_f)(const us_rtp_s *rtp);
us_rtp_s *us_rtp_init(unsigned payload, bool video, bool zero_playout_delay);
us_rtp_s *us_rtp_init(unsigned payload, bool video);
us_rtp_s *us_rtp_dup(const us_rtp_s *rtp);
void us_rtp_destroy(us_rtp_s *rtp);

2
janus/src/rtpa.c
View File

@ -26,7 +26,7 @@
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, false);
rtpa->rtp = us_rtp_init(111, false);
rtpa->callback = callback;
return rtpa;
}

11
janus/src/rtpv.c
View File

@ -31,10 +31,10 @@ void _rtpv_process_nalu(us_rtpv_s *rtpv, const uint8_t *data, size_t size, uint3
static ssize_t _find_annexb(const uint8_t *data, size_t size);
us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback, bool zero_playout_delay) {
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, zero_playout_delay);
rtpv->rtp = us_rtp_init(96, true);
rtpv->callback = callback;
return rtpv;
}
@ -59,12 +59,11 @@ char *us_rtpv_make_sdp(us_rtpv_s *rtpv) {
"a=rtcp-fb:%u nack pli" RN
"a=rtcp-fb:%u goog-remb" RN
"a=ssrc:%" PRIu32 " cname:ustreamer" RN
"%s" // playout-delay
"a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN
"a=sendonly" RN,
PAYLOAD, PAYLOAD, PAYLOAD, PAYLOAD,
PAYLOAD, PAYLOAD, PAYLOAD,
rtpv->rtp->ssrc,
(rtpv->rtp->zero_playout_delay ? "a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN : "")
rtpv->rtp->ssrc
);
return sdp;
# undef PAYLOAD
@ -78,6 +77,8 @@ void us_rtpv_wrap(us_rtpv_s *rtpv, const us_frame_s *frame) {
assert(frame->format == V4L2_PIX_FMT_H264);
rtpv->rtp->zero_playout_delay = (frame->gop == 0);
const uint32_t pts = us_get_now_monotonic_u64() * 9 / 100; // PTS units are in 90 kHz
ssize_t last_offset = -_PRE;

2
janus/src/rtpv.h
View File

@ -43,7 +43,7 @@ typedef struct {
} us_rtpv_s;
us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback, bool zero_playout_delay);
us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback);
void us_rtpv_destroy(us_rtpv_s *rtpv);
char *us_rtpv_make_sdp(us_rtpv_s *rtpv);