aler9
|
0c4f6e2d43
|
rtmp server: fix bias error in AAC DTS
|
3 years ago |
|
aler9
|
b6b99b142a
|
hls muxer: prefer hls.js over native HLS
|
3 years ago |
|
aler9
|
901eae2f6b
|
fix bias error in AAC timestamp
|
3 years ago |
|
aler9
|
58e3fa358e
|
split handling of on-demand sources and on-demand publishers
|
3 years ago |
|
aler9
|
98b3538289
|
fix panic that happens when publishing to a path with source = redirect (#933)
|
3 years ago |
|
aler9
|
35b3541e4f
|
hls client: add limit on AU size
|
3 years ago |
|
aler9
|
709d727eab
|
hls muxer: tune hls.js parameters
|
3 years ago |
|
aler9
|
dedca93eca
|
hls muxer: update hls.js
|
3 years ago |
|
aler9
|
ae7e68c914
|
hls muxer: remove progressive flag; add liveSyncDurationCount, liveMaxLatencyDurationCount to hls.js
|
3 years ago |
|
aler9
|
1e07636f86
|
change default RTSPS port (#867)
|
3 years ago |
|
aler9
|
6b86607092
|
rtsp source: improve support for AAC tracks with custom parameters
|
3 years ago |
|
aler9
|
f71b7d8967
|
fix tests
|
3 years ago |
|
aler9
|
ce42c53a03
|
hls, rtmp: fix video/audio sync
|
3 years ago |
|
aler9
|
f620484757
|
rtmp: always send decoder config before IDR frames
|
3 years ago |
|
aler9
|
98c6cd4650
|
RTSP: automatically remux oversized RTP/H264 packets; drop parameter ReadBufferSize
|
3 years ago |
|
aler9
|
58b2e7d24f
|
move trackID into data
|
3 years ago |
|
aler9
|
2c485f918b
|
fix tests
|
3 years ago |
|
aler9
|
dffe63f1bc
|
add SPS and PTS before IDRs of all incoming H264 streams; stop filtering H264 inside single protocols
|
3 years ago |
|
aler9
|
a34a01ebd9
|
RTMP client/source: support dynamic H264 SPS/PPS
|
3 years ago |
|
aler9
|
4d6f8b9b9b
|
RTSP client/source: support dynamic H264 SPS/PPS
|
3 years ago |
|
aler9
|
d929197b21
|
propagate H264 packets throughout the server
|
3 years ago |
|
aler9
|
a59ddf7176
|
rtsp server: remove useless check
|
3 years ago |
|
aler9
|
0605a2f369
|
update linter
|
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
|
56338e0084
|
hls client: do not create audio track when there's no audio track
|
3 years ago |
|
aler9
|
28063a1fbe
|
rename stream.onPacketRTP/RTCP into stream.writePacketRTP/RTCP
|
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
|
b4649ef60b
|
remove runOnPublish (breaking change)
|
3 years ago |
|
aler9
|
87f24f1704
|
update gortsplib
|
4 years ago |
|
aler9
|
507afbf73d
|
make logs more clear
|
4 years ago |
|
aler9
|
16df033d21
|
update gortsplib
|
4 years ago |
|
aler9
|
fe32022edf
|
hls client: move RTP packet generation outside client
|
4 years ago |
|
aler9
|
43471a05ab
|
update gortsplib
|
4 years ago |
|
aler9
|
5504ff44b6
|
rtsp: parse all incoming RTP/RTCP packets
|
4 years ago |
|
aler9
|
983469a1f9
|
rtmp: support clients that publish with empty metadata or no metadata (#386) (#769)
|
4 years ago |
|
aler9
|
f6a5fe2623
|
rtsp source: fix memory leak in case source doesn't send H264 params in time
|
4 years ago |
|
aler9
|
9c38b42b4c
|
fix crash that happened when publishing audio with RTMP (#810)
|
4 years ago |
|
aler9
|
3e8668f9e2
|
rtsp server: allow again H264 tracks without SPS or PPS in the SDP (#787)
|
4 years ago |
|
aler9
|
343a5f17fb
|
hls: add new parameter hlsSegmentMaxSize
|
4 years ago |
|
aler9
|
2bfdcc7d89
|
update gortsplib
|
4 years ago |
|
aler9
|
9735a8522e
|
fix shutdown freeze introduced by ef255af
|
4 years ago |
|
aler9
|
4a6d052bb7
|
add more debug log messages
|
4 years ago |
|
aler9
|
d3bf643f77
|
save regexp groups in G1, G2 env variables instead of 1, 2 (#642)
|
4 years ago |
|
aler9
|
c208cb9aff
|
update external authentication tests
|
4 years ago |
ShiBen
|
e3f63a43c9
|
External authentication support send url raw query
Do some dynamic authentication, such as token
|
4 years ago |
|
aler9
|
ef255af093
|
rtsp source: fix memory leak
This happened when the server was able to connect to the source,
but initialization failed before or during the PLAY request.
|
4 years ago |
|
aler9
|
b48e2f1f1b
|
add TODO comments
|
4 years ago |
|
aler9
|
49449eb5ad
|
Add new parameter 'runOnReady' (#752)
This is called when a stream is ready, whether it is published or proxied.
It replaces 'runOnPublsh'.
|
4 years ago |
