aler9
|
9bd8b2cfb6
|
rtmp server, hls muxer: fix DTS in case of B-frames and remove PTS-DTS offset
|
3 years ago |
|
aler9
|
4073013f68
|
hls muxer: stop normalizing PTS
|
3 years ago |
|
aler9
|
acd788d632
|
update gortsplib
|
3 years ago |
|
aler9
|
e7f88bc12f
|
move DTS extractor into gortsplib
|
3 years ago |
|
aler9
|
93975552ac
|
hls muxer: improve performance, decrease memory consumption
|
3 years ago |
Alessandro Ros
|
e115983296
|
Implement Low-Latency HLS (#938)
* add hlsVariant parameter
* hls: split muxer into variants
* hls: implement fmp4 segments
* hls muxer: implement low latency mode
* hls muxer: support audio with fmp4 mode
* hls muxer: rewrite file router
* hls muxer: implement preload hint
* hls muxer: add various error codes
* hls muxer: use explicit flags
* hls muxer: fix error in aac pts
* hls muxer: fix sudden freezes with video+audio
* hls muxer: skip empty parts
* hls muxer: fix video FPS
* hls muxer: add parameter hlsPartDuration
* hls muxer: refactor fmp4 muxer
* hls muxer: fix CAN-SKIP-UNTIL
* hls muxer: refactor code
* hls muxer: show only parts of last 2 segments
* hls muxer: implementa playlist delta updates
* hls muxer: change playlist content type
* hls muxer: improve video dts precision
* hls muxer: fix video sample flags
* hls muxer: improve iphone audio support
* hls muxer: improve mp4 timestamp precision
* hls muxer: add offset between pts and dts
* hls muxer: close muxer in case of error
* hls muxer: stop logging requests with the info level
* hls muxer: rename entry into sample
* hls muxer: compensate video dts error over time
* hls muxer: change default segment count
* hls muxer: add starting gap
* hls muxer: set default part duration to 200ms
* hls muxer: fix audio-only streams on ios
* hls muxer: add playsinline attribute to video tag of default web page
* hls muxer: keep mpegts as the default hls variant
* hls muxer: implement encryption
* hls muxer: rewrite dts estimation
* hls muxer: improve DTS precision
* hls muxer: use right SPS/PPS for each sample
* hls muxer: adjust part duration dynamically
* add comments
* update readme
* hls muxer: fix memory leak
* hls muxer: decrease ram consumption
|
3 years ago |
|
aler9
|
6de8277cb7
|
support SDPs with TIAS bandwidths (#948)
|
3 years ago |
|
aler9
|
2fba878755
|
rtsp server: fix wrong RTP-Info header (#922)
- fix truncation to seconds of RTP time in RTP-Info
- add a small quantity to rtptime in RTP-Info
- add 1 to last sequence number in RTP-Info
|
3 years ago |
|
aler9
|
7ecd52808a
|
rtsp source: support cameras that provide base URL inside control attribute (#683)
|
3 years ago |
|
aler9
|
d6a804f592
|
rtsp server: improve performance when reading with TCP
|
3 years ago |
|
aler9
|
f521049e2f
|
rtsp source: restore support for cameras with percent sign in password
|
3 years ago |
|
aler9
|
901eae2f6b
|
fix bias error in AAC timestamp
|
3 years ago |
|
aler9
|
1ad0df128d
|
rtsp source: fix timeout (https://github.com/aler9/rtsp-simple-server/discussions/897)
that happens when there's a UDP->TCP switch and Content-Base contains
an absolute URL
|
3 years ago |
|
aler9
|
6c95ea2837
|
rtsp source: support AUs wrapped inside ADTS (#855)
|
3 years ago |
|
aler9
|
411fb989eb
|
hls client: add limit on NALU size
|
3 years ago |
|
aler9
|
35b3541e4f
|
hls client: add limit on AU size
|
3 years ago |
|
aler9
|
91ce2dd886
|
rtsp server: fix reading with multicast when host is macOS (#895)
|
3 years ago |
|
aler9
|
c700e309ce
|
rtsp server: fix reading with ffplay and stateless ipv6
|
3 years ago |
|
aler9
|
0912042fb1
|
hls muxer: do not increase version number of MPEG-TS tables unless there's a reason (https://github.com/asticode/go-astits/pull/38)
|
3 years ago |
|
aler9
|
9098644802
|
rtsp source: fix parsing of transport header with dahua rtsp server (https://github.com/aler9/gortsplib/issues/121)
|
3 years ago |
|
aler9
|
385e301728
|
rtsp server/source: support AAC tracks with indexlength=0
|
3 years ago |
|
aler9
|
6b86607092
|
rtsp source: improve support for AAC tracks with custom parameters
|
3 years ago |
|
aler9
|
db7ee22789
|
rtsp source: support AAC tracks with custom sizelength, indexlength and indexdeltalength
(https://github.com/aler9/gortsplib/pull/118)
|
3 years ago |
|
aler9
|
2a0dc3b63d
|
rtsp: improve performance; add upper limit on NALU/AU size to prevent RAM exhaustion
|
3 years ago |
|
aler9
|
98c6cd4650
|
RTSP: automatically remux oversized RTP/H264 packets; drop parameter ReadBufferSize
|
3 years ago |
|
aler9
|
50d1079f70
|
RTSP: fix multicast TTL; set default multicast TTL to 16 (#873)
|
3 years ago |
|
aler9
|
d929197b21
|
propagate H264 packets throughout the server
|
3 years ago |
|
aler9
|
3fc4ca6465
|
update gortsplib; downgrade pion/rtp to v1
|
3 years ago |
|
aler9
|
f53b316c0d
|
rtsp server: generate RTCP sender reports automatically; stop routing RTCP packets
|
3 years ago |
|
aler9
|
a6986e9fa4
|
update gortsplib
|
3 years ago |
|
aler9
|
8cb2a7ffd4
|
rtsp source: support servers that sends UDP packets with a different IP (https://github.com/aler9/gortsplib/pull/108)
|
3 years ago |
|
aler9
|
544c192103
|
update gortsplib
|
3 years ago |
|
aler9
|
e611c7ddfd
|
RTSP source: skip tracks with an invalid clock rate (#832)
|
3 years ago |
|
aler9
|
245d58ee47
|
RTSP server: set timeout in Session header only when client is playing and protocol is UDP
|
3 years ago |
|
aler9
|
6dc11c2906
|
RTSP server/source: fix encoding of RTP packets with padding
This fixes a SIGSEGV with GStreamer.
|
3 years ago |
|
aler9
|
20a8862b2f
|
RTSP source: restore packet buffering with the UDP transport
|
3 years ago |
|
aler9
|
87f24f1704
|
update gortsplib
|
4 years ago |
|
aler9
|
d7edf42c40
|
rtsp server: restore authentication (#840)
|
4 years ago |
|
aler9
|
5f9b705a03
|
RTSP server: remove Session header from ANNOUNCE responses (#840)
|
4 years ago |
|
aler9
|
16df033d21
|
update gortsplib
|
4 years ago |
|
aler9
|
43471a05ab
|
update gortsplib
|
4 years ago |
|
aler9
|
5504ff44b6
|
rtsp: parse all incoming RTP/RTCP packets
|
4 years ago |
|
aler9
|
ae7b1c2ea6
|
rtmp, hls: fix wrong timestamp computation that happened at 28h28m
this was caused by a int64 overflow.
|
4 years ago |
|
aler9
|
7c1d329d61
|
rtsp server: fix multicast listen error on Windows (#742)
This happened when the system has network interfaces that don't support multicast.
|
4 years ago |
|
aler9
|
6c7fd8ee6f
|
fix bad track cloning and restore HLS audio support (#808)
|
4 years ago |
|
aler9
|
f330cc6997
|
rtsp source: fix crash in case a PLAY request fails
|
4 years ago |
|
aler9
|
343a5f17fb
|
hls: add new parameter hlsSegmentMaxSize
|
4 years ago |
|
aler9
|
2bfdcc7d89
|
update gortsplib
|
4 years ago |
|
aler9
|
9bc364fa21
|
support H264 tracks that contains extra data in sprop-parameter-set
|
4 years ago |
|
aler9
|
b8bf747796
|
update gortsplib
|
4 years ago |
