3189 Commits (c2a667cc35c11d43a7b27e70d4a5cdc027202ff4)

Author	SHA1	Message	Date
winlin	c2a667cc35	For #1638, #307, add comments for api and enum	5 years ago
winlin	2f0b150588	For #1638, #307, accept srs_string_dumps_hex to app utility.	5 years ago
winlin	e905fce847	For #1638, #307, add comments in RTMP/RTP message for RTC.	5 years ago
winlin	60c8b37f05	For #1638, #307, merge john, add NACK suport, remove debug code, verbose log	5 years ago
xiaozhihong	2b56f9ee59	add NACK suport, remove debug code, verbose log	5 years ago
winlin	12e99f1897	For #1638, #307, use webrtc adapter.js for demo.	5 years ago
winlin	a42cf3ae8d	For #1638, #307, refactor rtc config.	5 years ago
winlin	51af2b4779	For #1638, #307, rtc conf support ENV.	5 years ago
xiaozhihong	6c88f58079	remove debug code, fix rtc.html	5 years ago
winlin	d21ef106a0	Merge branch 'rtc' of https://github.com/xiaozhihong/srs into xiaozhihong-rtc	5 years ago
xiaozhihong	768598a3bb	add kernel rtp packet	5 years ago
xiaozhihong	027d34bbd3	add rtp shared packet	5 years ago
winlin	a7c8980a60	For #1636, muxing sh, use sound_rate if aac sr not set.	5 years ago
winlin	7036f839d2	For #1636, add comments	5 years ago
winlin	aa20a04abb	For #1636, fix bug for mux AAC to ADTS, never overwrite by RTMP sampling rate. 4.0.13	5 years ago
winlin	c66f33ac87	Merge pull request #1636 from runner365/3.0release.srt.dev ... support all aac sample rate in srt2rtmp	5 years ago
xiaozhihong	6decdc7838	adjust code style, fix some bug, add rtc session timeout	5 years ago
winlin	1382337cb4	Merge SRS3	5 years ago
winlin	856ed5a971	For #1635, inotify watch ConfigMap for reload. 3.0.134	5 years ago
winlin	bb509d23e2	For #1635, inotify watch ConfigMap for reload. 3.0.133	5 years ago
winlin	88368ed18d	For #1635, inotify watch ConfigMap for reload. 3.0.132	5 years ago
winlin	940798a9c7	For #1635, inotify watch ConfigMap for reload. 3.0.131	5 years ago
runner365	f102a598b8	Merge branch 'develop' into 3.0release.srt.dev	5 years ago
runner365	b9195122cf	support all sample rate in aac	5 years ago
winlin	5e60c8f9cc	For #1635, inotify watch ConfigMap for reload. 3.0.130	5 years ago
winlin	9b663c070a	Fix #1635, support auto reaload config by inotify. 3.0.129	5 years ago
winlin	b33751e2c1	Merge SRS3	5 years ago
winlin	4b395f6e16	Fix #1630, disable cache for stream changing, and drop dup header. 3.0.128	5 years ago
winlin	ed41ca867b	For #1594, detect and disable daemon for docker. 3.0.127	5 years ago
winlin	c1e07d6a96	For #1634, always check status in thread loop. 3.0.126	5 years ago
winlin	6a765ab849	Merge SRS3	5 years ago
winlin	6c55fd3e4b	For #1634, refactor output with datetime for ingest/encoder/exec. 3.0.125	5 years ago
winlin	c61c2a939f	For #1634, refactor output with datetime for ingest/encoder/exec. 3.0.125	5 years ago
HuyaJohn	da72caf8b9	h264 packet done, chrome play well	5 years ago
winlin	c78595c1fa	For #1634, refactor encoder output, support timestamp variables.	5 years ago
winlin	66b194cd63	For #1634, refactor encoder process management.	5 years ago
winlin	927c0c8e39	For #1634, fix quit by accident SIGTERM while killing FFMPEG. 3.0.124	5 years ago
xiaozhihong	e831f3254a	some code	5 years ago
HuyaJohn	ff0e03800d	h264 rtp debuging	5 years ago
xiaozhihong	e2675109fb	fix rtp h264 packet bug	5 years ago
HuyaJohn	3ae510b843	rtp dispatch done, but video can not play in chrome	5 years ago
xiaozhihong	2f462775a0	rtp support	5 years ago
HuyaJohn	2e68c375e3	start coding flv->rtp	5 years ago
xiaozhihong	c62901a3ac	make code easy, wrap udp remux socket	5 years ago
winlin	5586c2a128	For #1612, refactor return value for RTSP initialize.	5 years ago
winlin	dcb0553cc8	For #1612, fix crash bug for RTSP. 4.0.12	5 years ago
winlin	12a74326e8	Merge pull request #1612 from xialixin/develop ... Fix disconnect RTSP connection has assertion, resulting in program exit	5 years ago
winlin	41acf9ca8e	For #1631, support sei_filter for SRT. 4.0.11	5 years ago
runner365	a2839aacdf	add h264 sei filter	5 years ago
HuyaJohn	b730458d51	add candidates ip list, add rtc_upload.html to test	5 years ago