mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-08-01 19:25:10 +00:00
Code Issues 1 Projects Releases Wiki Activity
17983 commits 44 branches 1506 tags 395 MiB
Commit graph

48 commits

Author SHA1 Message Date
Aleix Conchillo Flaqué 086cc58952 srtpdec: fix rtp/rtcp sink event memory leak 
https://bugzilla.gnome.org/show_bug.cgi?id=740451
 2014-11-20 13:32:39 -05:00
Arun Raghavan 193eeb1243 srtpdec: Trivial documentation fix 2014-10-30 18:45:04 +05:30
Youness Alaoui 3c80396424 srtpdec: Add support for RTP/RTCP mixing if data arrives on RTCP pad 
https://bugzilla.gnome.org/show_bug.cgi?id=734321
 2014-08-21 18:45:19 -04:00
Miguel París Díaz bf003caf9d gstsrtpenc: unref event when needed 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731768
 2014-06-18 15:54:08 +02:00
Wim Taymans 52d4b7de07 srtpenc: improve property comments 
Add a Note to the allow-repeat-tx property because it is potentially
insecure when used incorrectly.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731714
 2014-06-18 15:37:23 +02:00
Miguel París Díaz c1052066e8 gstsrtpenc: add allow-repeat-tx property 2014-06-18 15:33:00 +02:00
Aleix Conchillo Flaqué da30669589 srtp: add support for rollover counters and replay protection window size 
We add a new signal, get-rollover-counter, to the SRTP encoder. Given a
ssrc the signal will return the currently internal SRTP rollover counter
for the given stream.

For the SRTP decoder we have a new SRTP caps parameter "roc" that needs
to be set when a new SRTP stream is created for a given SSRC.

https://bugzilla.gnome.org/show_bug.cgi?id=726861
 2014-06-03 16:18:25 -04:00
Olivier Crête 0566be384d srtpdec: Don't reset stream if the key hasn't changed 2014-05-06 00:49:59 -04:00
Olivier Crête b2a6490626 srtpdec: Add "remove-key" action signal to remove a single key 
Removing all the keys can be a problem as it also resets the various
counters, so instead add an option to remove a single key.
 2014-05-06 00:49:59 -04:00
Olivier Crête 53a4e05261 srtpenc: Use %u in templates to match rtpbin 2014-05-06 00:49:22 -04:00
Olivier Crête 8c9b1ade36 srtpdec: Remove unused field 2014-05-06 00:49:15 -04:00
Olivier Crête 5d3ce2c611 srtpenc: Add config for the replay window size 2014-05-02 17:27:56 -04:00
Olivier Crête b51344651e srtpenc: Remove unused object members 2014-05-02 17:25:35 -04:00
Olivier Crête 820200e20e srtpenc: Remove useless structure 
Remove now useless structure
 2014-05-02 17:19:51 -04:00
Olivier Crête 2906fae146 srtpenc: Remove ssrc parsing from stream 
It's no longer used
 2014-05-02 17:19:51 -04:00
Olivier Crête 98f8bf46df srtpenc: Remove SSRC from signal, it's meaningless 2014-05-02 17:19:51 -04:00
Olivier Crête 5a38ae5a83 srtpenc: Set the SRTP key as a shared key for all SSRCs 
This is already how it was used.
 2014-05-02 17:19:51 -04:00
Olivier Crête 775e998dab srtpdec: Only clear streams if they have been created 2014-05-01 19:16:35 -04:00
Olivier Crête 494b8e110d srtpdec: Use object debug 2014-05-01 19:10:43 -04:00
Aleix Conchillo Flaqué 01c15547d4 srtpdec: fix assertion checking ssrc from rtcp packets 
rtcp_buffer_get_ssrc is called even with RTP buffers. this means we
might end up with an exception and not find any valid RTCP packet type
and thus hit GST_RTCP_TYPE_INVALID. we now take care of this.

https://bugzilla.gnome.org/show_bug.cgi?id=727512
 2014-04-12 05:08:20 +02:00
Vincent Penquerc'h 54704b6f5d srtp: remove dead assignment 
Coverity 1197707
 2014-04-09 16:55:30 +01:00
Wim Taymans 9adb1734cf srtpenc: unlock object lock before posting error 
Posting an error with the object lock will deadlock. Rework the code
a bit to make things more simple.
 2014-03-25 17:46:32 +01:00
Aleix Conchillo Flaqué a04fe75047 srtpenc: unref caps after gst_query_set_caps_result 
https://bugzilla.gnome.org/show_bug.cgi?id=725412
 2014-03-01 16:40:39 +01:00
Sebastian Dröge 15e0db90e8 srtpdec: Fix comparison compiler warning 
error: comparison of constant -1 with expression of type
'GstSrtpCipherType' is always false
 2014-02-08 19:59:43 +01:00
Miguel París Díaz e944195b64 srtpdec: Fix GstCaps memory leak 
https://bugzilla.gnome.org/show_bug.cgi?id=721382
 2014-01-03 10:46:29 +01:00
Aleix Conchillo Flaqué fefd021ee2 srtp: add 256-bit key support 
* ext/srtp/gstsrtp.[ch]: added GST_SRTP_CIPHER_AES_256_ICM to
  GstSrtpCipherType and new function cipher_key_size.

* ext/srtp/gstsrtpenc.c: maximum key size is now 46 characters (14 for
  the salt plus the key). If different ciphers are chosen for RTP and
  RTCP the maximum needed key size is expected.

* ext/srtp/gstsrtpdec.c: minor documentation updates.

https://bugzilla.gnome.org/show_bug.cgi?id=720434
 2013-12-13 17:51:18 -05:00
Olivier Crête b58ca88fb9 srtpenc: Set the first session flag to false once one is create 
Otherwise it always thinks it's the first session.

https://bugzilla.gnome.org/show_bug.cgi?id=711495
 2013-11-19 20:18:48 -05:00
Olivier Crête 2b75eb85c4 srtpdec: Make sure that stream-id/caps/segment are sent before buffers 
It may be possible that only one of the two sink pads is linked in that case,
the events need to be created from the other pad.
 2013-11-19 20:12:54 -05:00
Olivier Crête de1fb842e6 srtpdec: Pass is_rtcp to setcaps function 2013-11-19 19:10:06 -05:00
Olivier Crête 8c76d7c77e srtp: Implement SRTCP demultiplexing 
Separate multiplexed RTP and RTCP if they arrive
on the rtp_sink pad.
 2013-11-19 18:34:08 -05:00
Tim-Philipp Müller b8f9d674be srtpdec: fix buffers being dropped due to shadowed variable 
Spotted by gstblub@gmail.com.

https://bugzilla.gnome.org/show_bug.cgi?id=706091
 2013-08-16 19:48:03 +01:00
Olivier Crête d346e75606 srtp: Let packets through without touching them if no crypto or auth is set 2013-08-12 12:27:33 -04:00
Olivier Crête a10e800dfd srtpdec: Refactor code a bit 
Simplify the error handling case and get the packet push out of the if()
 2013-08-12 12:27:16 -04:00
Olivier Crête 04bd37dd67 srtpdec: Don't error on invalid packet, just drop it 2013-08-12 11:57:04 -04:00
Olivier Crête ad7ffe64a6 srtp: Don't require a key if both auth and cipher are null 2013-06-28 20:35:16 -04:00
Olivier Crête 47a6118fbc srtp: Move the enums to the common header 
The types are used in both the encoder and decoder
 2013-06-28 20:35:15 -04:00
Olivier Crête 4d7a186513 srtp: Remove unused code 2013-06-28 20:13:38 -04:00
Edward Hervey f88f1d0a35 srtp: Fix dist 
The enums should not be dist-ed and instead be re-generated when
compiling.
 2013-06-23 09:06:29 +02:00
Sebastian Dröge e51cd4fe2f gst: Add better support for static plugins 2013-04-15 15:59:22 +02:00
Olivier Crête 4a9c3d4f16 srtpdec: Don't stop on invalid buffer 2013-04-09 21:37:25 -04:00
Olivier Crête 3bf994eb3b srtpdec: Cleaner stream cleanup 2013-04-09 21:37:25 -04:00
Olivier Crête 4b89870a58 srtpdec: Don't crash on invalid cipher 2013-04-09 21:37:24 -04:00
Tim-Philipp Müller 5beefc66cd ext: printf format fixes in debug and error messages 2013-04-08 17:40:14 +01:00
Olivier Crête c42ac5b450 srtp: Update example pipelines 2013-03-28 14:35:48 -04:00
Olivier Crête ed7f675966 srtp: Fix filter handling in caps negotiation to keep filter content 2013-03-25 13:48:56 -04:00
Jose Antonio Santos Cadenas ea2332c466 srtp: fix compilation warning returning NULL in case of error 
https://bugzilla.gnome.org/show_bug.cgi?id=695897
 2013-03-15 11:04:39 +00:00
Jose Antonio Santos Cadenas 2347c4e711 srtp: use constant AES_128_ICM instead of AES_ICM that is not found 
https://bugzilla.gnome.org/show_bug.cgi?id=695897
 2013-03-15 11:04:29 +00:00
Olivier Crête 4eb960d6c3 srtp: Add SRTP plugin based on libsrtp 
In large part the work of Gabriel Millaire
 2013-03-12 20:01:50 -04:00