aler9
|
e45820b2c0
|
hls server: show real client IPs when behind a proxy (#955)
|
3 years ago |
|
aler9
|
3e5f62156d
|
fix DTS error in case of H264 NALUs without POC
|
3 years ago |
|
aler9
|
05bac43177
|
rtmp: fix compatibility with some dji drones (#928)
|
3 years ago |
|
aler9
|
cb610a707c
|
hls muxer: fix code 500 when a stream is not found
|
3 years ago |
|
aler9
|
9d3fd3bc37
|
update gortsplib
|
3 years ago |
|
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
|
2ed1aa3d11
|
hls muxer, rtmp server: extract DTS from samples
|
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
|
c5afa69174
|
fix regression that caused a source to be closed after 10secs when sourceOnDemand is yes (#949)
|
3 years ago |
|
aler9
|
23ac079646
|
rtsp source: fix regression (#949)
This caused a periodic disconnection when sourceOnDemand is yes
|
3 years ago |
|
aler9
|
13fb24da39
|
fix rtmp -> rtsp audio conversion (#932)
|
3 years ago |
|
aler9
|
d6a804f592
|
rtsp server: improve performance when reading with TCP
|
3 years ago |
|
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
|
3 years ago |
|
aler9
|
507afbf73d
|
make logs more clear
|
3 years ago |
|
aler9
|
16df033d21
|
update gortsplib
|
3 years ago |
|
aler9
|
fe32022edf
|
hls client: move RTP packet generation outside client
|
3 years ago |
|
aler9
|
43471a05ab
|
update gortsplib
|
3 years ago |
|
aler9
|
5504ff44b6
|
rtsp: parse all incoming RTP/RTCP packets
|
3 years ago |
