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* fs-rtp-substream: Drop non-serialized events without capsOlivier Crête2016-03-101-0/+4
| | | | | This prevents some events that shouldn't be forwarded from going downstream.
* rtpstream: Accept all uncrypted packets if no crypto setOlivier Crête2015-07-243-1/+36
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* rtpstream: Accept everything when no crypto was setOlivier Crête2015-07-241-4/+2
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* bitrateadapter: Template caps are ANY and absorb the rest, so ignore themOlivier Crête2015-04-271-6/+0
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* rtp: Opus is now our favorite codecOlivier Crête2015-03-251-0/+2
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* rtp: Parse payloaders with multiple namesOlivier Crête2015-03-251-0/+5
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* rtp: Put channels as encoding-params as expectedOlivier Crête2015-03-251-2/+9
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* fsrtpxdata: Add gst-plugins-base libs to the CFLAGS and LIBSSebastian Dröge2015-03-191-0/+2
| | | | | Fixes compilation in gst-uninstalled, as otherwise the RTP library is not found.
* Enable building static GStreamer pluginsNicolas Dufresne2015-02-255-0/+5
| | | | https://bugs.freedesktop.org/show_bug.cgi?id=89287
* stream: Add "require-encryption" parameterOlivier Crête2015-02-252-11/+48
| | | | | | | If it is set to TRUE, then all buffers will be dropped before the decryption key is set. https://bugs.freedesktop.org/show_bug.cgi?id=89288
* rtpconference: Make get_extension() staticOlivier Crête2015-02-041-1/+1
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* rtpxdatapay: Use gst_buffer_copy_into to avoid unreffing the bufferYouness Alaoui2015-01-271-1/+3
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* rtpxdatapay: Add support for MTU and split long messages into multiple packetsYouness Alaoui2015-01-271-5/+32
| | | | | | Split all messages into max 1200 bytes of payload and send a GstBufferList when needed Keep sending a normal buffer in case the buffer is smaller than 1200 bytes to things slightly faster
* Add support for send-rtcp-mux on fs-rtp-session and nice transmitterYouness Alaoui2015-01-271-1/+42
| | | | | | In fs_nice_transmitter_set_send_component_mux(), the component IDs, which start from 1, are used as nicesinks array indexes and nicesinks[0] is always NULL.
* msnconnection: Fix potential race/deadlockOlivier Crête2015-01-271-1/+5
| | | | Unlock the mutex while waiting for the thread to exit.
* Fix clang warningsOlivier Crête2014-10-281-1/+1
| | | | https://bugs.freedesktop.org/show_bug.cgi?id=85565
* rtp-discover-codecs: Make global access to blueprints thread-safeNicolas Dufresne2014-10-281-4/+17
| | | | | | | | The global variable list_codec_blueprintfs refcounted with codecs_list_ref was not thread safe. This patch uses a global lock to make this code path thread safe. https://bugs.freedesktop.org/show_bug.cgi?id=85567
* rtpsession: Don't try to return srtpenc/dec if not installedOlivier Crête2014-10-281-2/+2
| | | | https://bugs.freedesktop.org/show_bug.cgi?id=85566
* rtpstream: Use the right variables in validationOlivier Crête2014-10-091-5/+5
| | | | Copy-paste error
* rtpsession: Don't try to start sending before a transmitter is setOlivier Crête2014-10-071-9/+4
| | | | No stream is really sending before a transmitter is set.
* session: Add internal-session propertyYouness Alaoui2014-09-181-1/+13
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* rtpsession: Fix discovery of RTCP ssrc.Youness Alaoui2014-09-181-16/+27
| | | | | | SRTCP packets will have SDES encrypted, so we need to check for RR and SR reports. Also, the code was checking if rtcp_map failed, instead of succeeded. This also allows us to mix rtp and rtcp on the same component.
* rtp: Implement setting SRTP decryption keyOlivier Crête2014-09-183-59/+302
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* rtpsession: Factor out SRTP parameter validationOlivier Crête2014-09-183-163/+192
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* rtpsession: Implemnt setting SRTP encryption keyOlivier Crête2014-09-181-1/+278
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* rtpsession: Create srtpenc & srtpdecOlivier Crête2014-09-181-33/+143
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* transmitter: Remove recvonly-filterOlivier Crête2014-09-181-9/+0
| | | | It was pretty much useless anyway.
* Remove fsrtcpfilterOlivier Crête2014-09-184-333/+0
| | | | It's not useful in real life
* fsrtpxdata: Add RTP pay/depay for Microsoft Lync RTP x-dataOlivier Crête2014-09-186-0/+399
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* Add Application Media typeOlivier Crête2014-09-184-1/+24
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* rtpdiscocodec: Also discover formats with no encoderOlivier Crête2014-09-181-5/+16
| | | | This makes L16, L24 and Video/RAW available to Farstream without profiles.
* rtpdiscocode: Ignore codecs with no "payload" propertyOlivier Crête2014-09-181-0/+2
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* rtpdiscocodec: Use GQueue instead g_list_appendOlivier Crête2014-09-181-3/+3
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* rtpbitrateadapter: Check bitrate changes since last getcaps or last notificationOlivier Crête2014-09-182-12/+10
| | | | Not only since the last reconfigure
* rtpbitrateadapter: Remove unused capsOlivier Crête2014-09-182-8/+1
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* rtpbitrateadapter: Support non-video caps tooOlivier Crête2014-09-181-3/+2
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* rtpbitrateadapter: Do dynamic getcapsOlivier Crête2014-09-181-9/+36
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* rtpbitrateadapter: Let upstream do possible renegotiation on > 10% bitrate ↵Olivier Crête2014-09-182-53/+28
| | | | change
* rtpbitrateadapter: Remove caps propertyOlivier Crête2014-09-181-39/+0
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* rtpbitrateadapter: Pass media type from callerOlivier Crête2014-09-181-15/+18
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* rtpbitrateadapter: Remove gray capsOlivier Crête2014-09-181-33/+15
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* rtpcodecnego: Filter by input and output capsOlivier Crête2014-09-183-3/+66
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* rtpsession: Implement allowed caps settingOlivier Crête2014-09-181-1/+83
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* rtpsession: Add input and output capsOlivier Crête2014-09-181-0/+15
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* rtpcodecnego: Skip full list iterationOlivier Crête2014-09-181-0/+3
| | | | One send codec is enough to be happy!
* rtpcodecnego: Discover input/output caps for application specified pipelinesOlivier Crête2014-09-184-29/+53
| | | | | From the codec preferences, if there is a pipeline, inspect to find the possible caps.
* rtpdiscocodec: Factor out in/out caps discoveryOlivier Crête2014-09-181-86/+96
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* rtpcodecnego: Add CodecPreference struct wrapping codec prefsOlivier Crête2014-09-183-59/+89
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* rtp: Use FsStreamDirection instead of "is_send" gbooleanOlivier Crête2014-09-186-46/+92
| | | | Makes the API clearer
* rtpcodecdisco: Discover output caps from receive codecbinOlivier Crête2014-09-181-4/+33
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