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* rtp: stop the transmitter src before unlinking its funnelFabrice Bellet2018-10-311-7/+24
| | | | | | | | | | | This patch tweaks the order the elements are stopped and unlinked to prevent the transmitter source to fail on a not-linked to any sinkpads error. The pipeline is transmitter-src -> funnel -> rtpbin -> substream. The funnel is stopped, then the transmitter-src, and thereafter the funnel is unlinked. https://bugs.freedesktop.org/show_bug.cgi?id=100586
* rtp: fix a double locking issue on the sessionFabrice Bellet2018-10-313-7/+12
| | | | | | | | | The session value used in fs_rtp_stream_add_substream_unlock(), taken from the stream struct may be null, while the session value from fs_rtp_session_new_recv_pad() is not. However these two function depend on the same session value to properly lock and unlock it: the first function will unlock the session previously locked by the second function.
* rtpconference: Move link flags to convenience libraryNicolas Dufresne2017-06-211-7/+8
| | | | | | This way unit test will inherit from all the required flags. https://bugs.freedesktop.org/show_bug.cgi?id=101544
* fsrtpsession: Set discovery valve to dropping on creationSergey Mamonov2017-06-061-0/+2
| | | | | | Although it should do nothing, it seems to improve CPU usage. https://bugs.freedesktop.org/show_bug.cgi?id=100412
* rtptfrc: Fix reference countingFabrice Bellet2017-06-051-2/+4
| | | | https://bugs.freedesktop.org/show_bug.cgi?id=99823
* stream: Stop substreams before removing themOlivier Crête2017-06-051-0/+1
| | | | https://bugs.freedesktop.org/show_bug.cgi?id=100644
* rtp: test the session conference property before using itFabrice Bellet2017-06-051-0/+20
| | | | | | | | | This may happen when the rtp session object is calling its dispose function in another thread. The disposed flag is set, and it prevents the fs_rtp_session_get_property() function to return its conference object. https://bugs.freedesktop.org/show_bug.cgi?id=101169
* Remove MSN pluginOlivier Crête2017-01-1715-3768/+0
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* rtpbitrateadapter: should make no adaption by defaultJakub Adam2016-12-221-2/+4
| | | | | | | | | | | | | | | | Description of "bitrate" property says 0 (the default value) means the element performs no adaption, and so one would assume it would remain passive until "bitrate" is set to some nonzero value. However, when "bitrate" is left unset, the adapter instead requests video in tiny 128x96 resolution on its sink pad. In order for fs_rtp_bitrate_adapter_getcaps() to return peer_caps by default, the value of FsRtpBitrateAdapter::bitrate has to be initialized to G_MAXUINT. Also fix the comments to say that MAXUINT is no adaptation. https://bugs.freedesktop.org/show_bug.cgi?id=99183
* rtp: Switch VP8 to standard encoding nameOlivier Crête2016-12-181-1/+1
| | | | | | | | This has been changed in GStreamer a very long time ago. Issue reported by Fabrice Bellet https://bugs.freedesktop.org/show_bug.cgi?id=99122
* rtp-codec-specific: Add OPUS non-negotiationOlivier Crête2016-07-141-1/+13
| | | | Also include unit test
* rtp-codec-specific: Document types betterOlivier Crête2016-07-141-3/+5
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* rtp-tfrc: Fix memset to the right sizeOlivier Crête2016-06-161-2/+2
| | | | | | This was reported from static analysis by dcb314@hotmail.com https://bugs.freedesktop.org/show_bug.cgi?id=96546
* 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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