bc3084ae7b
support proxying WebRTC streams ( #2142 )
2 years ago
e3d4856b4f
update gortsplib ( #2126 )
2 years ago
db3862cf0d
move stream in a dedicated package ( #2121 )
...
needed by #2068
2 years ago
39c072edd6
change repository owner ( #1801 )
2 years ago
596765c14c
Allow sending Range header to RTSP sources ( #1780 )
...
* Enable Range headers using path config
* Use enum instead of strings
* Comments added to new code
* Wrong comment format
* Made CreateRangeHeader func private
* reorder configuration
* handle errors inside createRangeHeader()
* add tests
* update API docs
---------
Co-authored-by: Jordy Boezaard <jordy@boezaard.com>
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
2 years ago
1688e5d2e5
support publishing with WebRTC ( #1659 ) ( #1786 )
2 years ago
225220ddd5
print warning in case no key frames are being received ( #1763 )
2 years ago
2d17dff3b5
support publishing, reading and proxying MPEG-2 audio (MP3) tracks with RTMP ( #1102 ) ( #1736 )
2 years ago
efb6060b45
update gortsplib ( #1671 )
2 years ago
053f2ec282
rename repository and executable ( #1641 )
2 years ago
2dffccf9c1
update gortsplib, gohlslib ( #1637 )
2 years ago
51a1cb9c78
update gohlslib ( #1583 )
2 years ago
e8bdad8a1e
rename Data into Unit ( #1556 )
2 years ago
b461f15a4c
rpicamera: support changing parameters without interrupting the stream ( #1463 )
3 years ago
e3fff72b7c
move format processors into dedicated folder
3 years ago
034e42f463
hls muxer: support reading Opus tracks ( #1338 )
3 years ago
5efe97abf1
Support reading with WebRTC ( #1242 )
3 years ago
4bafa4ea9b
add dedicated processors for H265 and Opus
3 years ago
c778c049ce
switch to gortsplib v2 ( #1301 )
...
Fixes #1103
gortsplib/v2 supports multiple formats inside a single track (media). This allows to apply the resizing algorithm to single formats inside medias.
For instance, if a media contains a a proprietary format and an H264 format, and the latter has oversized packets, they can now be resized.
3 years ago
e605727c78
produce same absolute time in RTSP and HLS ( #1249 )
...
* add a NTP timestamp to each data unit
* use that NTP timestamp in all protocols
3 years ago
282d155a4f
update gortsplib
3 years ago
0943b269ab
Decode streams once and only when needed ( #1218 )
...
* split data into specialized structs
* move MPEG4-audio decoding into streamTrack
* restore video/audio synchronization in HLS muxer and RTMP server
* log decode errors
* move H264 decoding and re-encoding here from gortsplib
* add tests
* update gortsplib
3 years ago
f0514b3983
update gortsplib
3 years ago
584cc9357a
rtsp source: log decode errors
3 years ago
8c2b12a314
Add native support for the raspberry pi camera ( #1057 )
3 years ago
cb8aec8844
print track codecs into logs
3 years ago
f952ff25f6
remove useless append()
3 years ago
3606472e82
generate RTP packets after H264 remuxing
...
Previously, RTP packets coming from sources other than RTSP (that
actually are RTMP and HLS) were generated before the H264 remuxing, and
that leaded to invalid streams, expecially when sourceOnDemand is true
and the stream has invalid or dynamic SPS/PPS.
3 years ago
055e08ac6c
remove "on" prefix from most communication functions between components
3 years ago
7067c02030
api: return static sources in paths/list even if sourceOnDemand is true
3 years ago
6afbdf1015
add intermediate layer between path and static sources
3 years ago
34eb1d1e7a
normalize sources
3 years ago
9d3fd3bc37
update gortsplib
3 years ago
98c6cd4650
RTSP: automatically remux oversized RTP/H264 packets; drop parameter ReadBufferSize
3 years ago
58b2e7d24f
move trackID into data
3 years ago
d929197b21
propagate H264 packets throughout the server
3 years ago
0605a2f369
update linter
3 years ago
3fc4ca6465
update gortsplib; downgrade pion/rtp to v1
3 years ago
f53b316c0d
rtsp server: generate RTCP sender reports automatically; stop routing RTCP packets
3 years ago
a6986e9fa4
update gortsplib
3 years ago
28063a1fbe
rename stream.onPacketRTP/RTCP into stream.writePacketRTP/RTCP
3 years ago
6dc11c2906
RTSP server/source: fix encoding of RTP packets with padding
...
This fixes a SIGSEGV with GStreamer.
3 years ago
87f24f1704
update gortsplib
4 years ago
5504ff44b6
rtsp: parse all incoming RTP/RTCP packets
4 years ago
f6a5fe2623
rtsp source: fix memory leak in case source doesn't send H264 params in time
4 years ago
2bfdcc7d89
update gortsplib
4 years ago
9735a8522e
fix shutdown freeze introduced by ef255af
4 years ago
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
8ac665be87
un-capitalize private fields
4 years ago
a1fed6fb38
rtsp source: set ServerName when using TLS ( #708 )
4 years ago
Alessandro Ros
Jordy84
aler9