Martin Pulec
5c48277b10
BasicRTSPOnlySubsession: basic support for IPv6
...
\+ do not force IPv4 for RTSP
2024-07-19 12:03:27 +02:00
Martin Pulec
76e1e6cdcc
BasicRTSPOnlySubsession: fixed leaked memory
2024-07-19 12:03:27 +02:00
Martin Pulec
dd69d4fd83
rxtx/rtsp: print SDP lines in verbose
2024-07-19 12:03:13 +02:00
Martin Pulec
8adb8f2c50
current live555 compat
2024-07-19 12:01:52 +02:00
Martin Pulec
ace6d570c9
Revert "vrxtx/rtsp subsessions: do not send msgs on exit"
...
This reverts commit c3bb31928e
2024-02-08 14:49:49 +01:00
Martin Pulec
e8af3008d7
rtsp: support for mp3 - use get_audio_rtp_pt_rtpmap
2024-02-05 15:08:34 +01:00
Martin Pulec
c3bb31928e
vrxtx/rtsp subsessions: do not send msgs on exit
...
When destroying the state, ServerMediaSubsession::deleteStream callbacks
are called but we are already deconfiguring so do not send messages to
change destination (eg. audio is already destroyed so the message wont
be delivered causing warnings like:
```
Receiver audio.sender does not exist.
Receiver audio.sender does not exist.
Warning: Message queue not empty!
```
Currently this is just a dirty fix, rtsp_serv::watch would be better to
pass.
2024-01-15 16:35:30 +01:00
Martin Pulec
1528b504ea
RTSP: fixed leaked Destination on TEARDOWN/timeout
...
attributes were set to NULL without deleting the content
2024-01-09 16:20:58 +01:00
Martin Pulec
1ad7722a67
RTSP: support stream redirect
...
When the client doesn't call TEARDOWN (like ffplay doesn't),
the stream could not have been played until the timeout (given by
`reclamationTestSeconds`). After that (or when TEARDOWN was called),
`BasicRTSPOnlySubsession::deleteStream()` is called allowing the new
stream.
After this change, the stream can be redirected withot explicit TEARDOWN
or timeout.
2024-01-09 16:20:18 +01:00
Martin Pulec
64d088f5e7
SDP: set audio ch count 2 for Opus
...
According to RFC 7587, channel count must be set always to 2 (actual
channel count like mono is signalized in-band in Opus)
2024-01-08 17:18:31 +01:00
Martin Pulec
dbbb6d0963
fixed BasicRTSPOnlySubsession snprintf
...
Fixes the commit 580ac72e
2024-01-08 14:57:09 +01:00
Martin Pulec
48219758ea
use Opus, not OPUS
...
For audio codecs, we respect its native capitalization of letters, eg.
AAC, speex. So do it also for Opus. This should not affect existing
applications since the Opus name is parsed case-insensitively.
Only exception is SDP (rtpmap) where is usually used lower-case (at
least in rfc7587).
2023-04-26 09:53:10 +02:00
Martin Pulec
580ac72ec2
replaced all remaining sprintf witn snprintf
...
using bound checking variants
Remained last one instance in utils/text.c, that does the checking by
itself and vsnprintf compat using vsprintf, that is not used, anyways.
2023-04-13 14:04:29 +02:00
Martin Pulec
320ea5d3df
Fixed some Coverity warnings
2021-03-25 15:08:03 +01:00
Martin Pulec
4aaec8f7f0
Fixed some Coverity warnings
2021-02-23 15:00:58 +01:00
Martin Pulec
817fe39684
RTSP: correctly free responses
2020-02-10 08:14:52 +01:00
Martin Pulec
31d9809fcd
Updated documentation
...
Updated authors, copyright to 3-clause BSD (where possible) and file-level Doxygen
2019-11-09 13:47:11 +01:00
Martin Pulec
ea1971c116
RTSP: compatible with current UltraGrid
...
+ note about older version which is needed for live555
2017-05-19 15:57:49 +02:00
Martin Pulec
5f103459fa
Cleaned warnings
2015-12-14 17:01:04 +01:00
Martin Pulec
2cc6aab0e2
Added possibility to send message synchronously
...
+ in capabilities list, given bitrate is computed according to the
detected capture format (provided that '-t' argument is given)
2015-08-25 17:05:23 +02:00
Martin Pulec
754cec0ff2
RTSP server minor fix: delete sent message
2015-01-20 14:05:52 +01:00
Gerard CL
7a20e0b26f
merge for opus codec over rtp transmission support
2014-10-09 12:28:54 +02:00
Castillo, Gerard
360ac36373
adding opus standard support for rtsp server
2014-08-15 14:18:49 +02:00
Martin Pulec
71faa05ebb
RTSP: fixed type names (MSW compatibility)
2014-08-06 12:27:58 +02:00
Castillo, Gerard
240340b0e0
rtsp subsession code cleanup
2014-07-21 11:30:11 +02:00
Castillo, Gerard
93763d8b69
rtsp server defines working properly and small improvements
2014-07-21 08:57:41 +02:00
Gerard CL
e322ffdbde
standards - rtsp with new audio and video refactors and sync standard working
2014-05-20 16:51:35 +02:00
Castillo, Gerard
8f47dae291
rtsp already working with huge refactor from CESNET, now reparing audio messaging api issue
2014-03-10 15:09:18 +01:00
Castillo, Gerard
12eda9a77b
refactor for enumerating avtypes for rtsp accepted formats
2014-02-26 12:30:37 +01:00
Castillo, Gerard
88657442b5
rtsp server issue solved
2014-02-14 14:00:33 +01:00
Castillo, Gerard
e0fb6a3db7
rtsp server issue solved, now reinits with any possible request working
2014-02-12 09:54:50 +01:00
Castillo, Gerard
4826e146c6
audio and/or video RTSP server working
2014-01-10 13:04:29 +01:00
Castillo, Gerard
09ec6d7833
header files
2013-12-19 11:00:59 +01:00
Castillo, Gerard
2fbaba6af5
cleanups
2013-12-19 10:33:11 +01:00
Castillo, Gerard
09fb978293
rtsp server working and clean-up done
2013-12-17 13:14:27 +01:00
Castillo, Gerard
e7e637b410
first change port then addrs
2013-12-13 13:18:09 +01:00
Castillo, Gerard
27d983a625
RTSPOnlySubsession class improvements
2013-12-11 15:08:26 +01:00
Castillo, Gerard
1f5a34b5d0
rtsp classes
2013-12-11 08:58:52 +01:00
