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refactoring
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@@ -116,7 +116,9 @@ static void *_common_thread(void *v_client, bool video) {
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packet.extensions.max_delay = 0;
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} else {
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packet.extensions.min_delay = 0;
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packet.extensions.max_delay = 1000;
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// 10s - Chromium/WebRTC default
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// 3s - Firefox default
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packet.extensions.max_delay = 300; // == 3s, i.e. 10ms granularity
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}*/
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client->gw->relay_rtp(client->session, &packet);
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@@ -56,8 +56,8 @@
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#include "config.h"
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static us_config_s *_g_config = NULL;
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const useconds_t _g_watchers_polling = 100000;
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static us_config_s *_g_config = NULL;
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static const useconds_t _g_watchers_polling = 100000;
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static us_janus_client_s *_g_clients = NULL;
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static janus_callbacks *_g_gw = NULL;
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@@ -107,7 +107,8 @@ static void *_video_rtp_thread(UNUSED void *arg) {
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us_frame_s *frame;
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if (us_queue_get(_g_video_queue, (void **)&frame, 0.1) == 0) {
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_LOCK_VIDEO;
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us_rtpv_wrap(_g_rtpv, frame);
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const bool zero_playout_delay = (frame->gop == 0);
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us_rtpv_wrap(_g_rtpv, frame, zero_playout_delay);
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_UNLOCK_VIDEO;
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us_frame_destroy(frame);
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}
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@@ -71,13 +71,13 @@ char *us_rtpv_make_sdp(us_rtpv_s *rtpv) {
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#define _PRE 3 // Annex B prefix length
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void us_rtpv_wrap(us_rtpv_s *rtpv, const us_frame_s *frame) {
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void us_rtpv_wrap(us_rtpv_s *rtpv, const us_frame_s *frame, bool zero_playout_delay) {
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// There is a complicated logic here but everything works as it should:
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// - https://github.com/pikvm/ustreamer/issues/115#issuecomment-893071775
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assert(frame->format == V4L2_PIX_FMT_H264);
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rtpv->rtp->zero_playout_delay = (frame->gop == 0);
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rtpv->rtp->zero_playout_delay = zero_playout_delay;
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const uint32_t pts = us_get_now_monotonic_u64() * 9 / 100; // PTS units are in 90 kHz
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ssize_t last_offset = -_PRE;
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@@ -47,4 +47,4 @@ us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback);
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void us_rtpv_destroy(us_rtpv_s *rtpv);
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char *us_rtpv_make_sdp(us_rtpv_s *rtpv);
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void us_rtpv_wrap(us_rtpv_s *rtpv, const us_frame_s *frame);
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void us_rtpv_wrap(us_rtpv_s *rtpv, const us_frame_s *frame, bool zero_playout_delay);
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