GStreamer memo

by

https://gstreamer.freedesktop.org/documentation/tools/gst-launch.html

https://cloud.tencent.com/developer/article/1820526
https://blog.csdn.net/csdnhuaong/article/details/80026433
https://blog.csdn.net/qq_42711516/article/details/123921984
https://blog.csdn.net/han2529386161/article/details/102724856

https://blog.csdn.net/m0_51004308/article/details/121357638
https://blog.csdn.net/Aidam_Bo/article/details/109772430

gstreamer之RTSP Server一个进程提供多路不同视频
https://blog.51cto.com/u_13161667/3310768

全网首发：gstreamer如何接入RTSP流（IP摄像头）的代码范例
https://blog.51cto.com/u_13161667/3310521

GstMP4Mux caps
DeepStream/Gstreamer queue实现预缓存功能
https://blog.csdn.net/qq_41632852/article/details/124959601

gstreamer中tee如何实现动态增减支路（预览+截图+录像）
https://blog.csdn.net/qq_41563600/article/details/121343927

【gstreamer opencv::Mat】将opencv的cv::Mat数据转换成MP4视频
https://blog.csdn.net/weixin_44495869/article/details/121900517

#播放并存储为h264
gst-launch-1.0 -e -v  udpsrc port=10001 !  "application/x-rtp, media=video, clock-rate=90000, encoding-name=H264"  ! rtph264depay ! tee name=t  t. ! queue ! h264parse ! "video/x-h264, stream-format=byte-stream" ! filesink location=./aa.h264 t. !  queue ! avdec_h264 ! glimagesink

#混流+缩放
./gst-launch-1.0  intervideosrc !  video/x-raw,framerate=(fraction)10/1,width=3200,height=2400 ! videomixer name=mix sink_1::xpos=0 sink_2::xpos=1000 sink_2::width=100 sink_2::height=100 ! videoconvert ! glimagesink filesrc location=./ttt.mov ! decodebin ! video/x-raw,width=1280,height=720 ! videoscale ! video/x-raw,width=100  ! mix.  videotestsrc ! video/x-raw ! mix.

gst-launch-1.0.exe videotestsrc ! queue ! d3d11videosink

http://t.zoukankan.com/missmzt-p-10918216.html

GStreamer插件：appsrc 解决播放实时视频流高延迟问题
https://blog.csdn.net/zzs0829/article/details/111562443

https://gstreamer.freedesktop.org/documentation/tutorials/basic/debugging-tools.html

g_setenv("GST_DEBUG_DUMP_DOT_DIR", "D:/work/gstreamer/gstdot", true);
gst_init(NULL, NULL);

GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(pipeline), GST_DEBUG_GRAPH_SHOW_ALL, "capture1234");

g_setenv("GST_DEBUG_FILE", "D:/work/gstreamer/gstdot/output.log", true);
g_setenv("G_MESSAGES_DEBUG","all",TRUE);
gst_debug_set_default_threshold(GST_LEVEL_TRACE);

http://www.graphviz.org/doc/info/command.html

1
2
3
4

push
	upstream element calls downstream sink pads' gst_pad_push
pull
	downstream element calls upstream source pads' gst_pad_pull_range

gst_rtspsrc_loop_interleaved
gst_rtspsrc_stream_configure_manager
gst_qt_mux_sink_event_pre_queue

rtp_session_process_sr
https://stackoverflow.com/questions/53169699/gstreamer-calculate-delay-in-received-video-frames-buffers-to-detect-communicat