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* rtpbitrateadapter: Let upstream do possible renegotiation on > 10% bitrate ↵Olivier Crête2014-09-182-53/+28
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* 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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* rtpcodecdisco: Discover input caps from send codecbinOlivier Crête2014-09-181-26/+109
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* rtpcodeccache: Add input/output caps to the codec cacheOlivier Crête2014-09-183-2/+49
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* rtpsession: Only try to read RTCP if mapping succeededOlivier Crête2014-07-311-1/+1
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* rtpdiscocodec: Fix build with -Werror=format-securityJakub Adam2014-05-121-1/+1
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* rtpconference: Fix debug printOlivier Crête2014-05-091-2/+1
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* rtpcodecdisco: Style fixOlivier Crête2014-05-091-1/+1
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* rtpsession: Verify object exists in dispose pathOlivier Crête2014-05-091-1/+2
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* rtpdiscocodec: Debug pipeline at the right levelOlivier Crête2014-05-091-1/+1
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* rtpcodecdisco: Use GST_PTR_FORMAT for caps debugOlivier Crête2014-05-071-7/+2
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* rtpcodecdisco: Improve debug printingOlivier Crête2014-05-071-13/+32
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* videoanyrate: Use the Fs prefix internally tooOlivier Crête2014-05-062-31/+30
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* rtp: Fix a couple typosVincent Penquerc'h2014-05-061-2/+2
| | | | https://bugs.freedesktop.org/show_bug.cgi?id=37997
* Fix some mismatched / redundant <para> gtk-doc tagsIain Lane2014-05-061-2/+2
| | | | | | | | | gtk-doc 1.20 got more strict about correctness here wrt. balanced tags. Also bump the relevant gtk-doc requirement, these tags were added for a reason back then. https://bugs.freedesktop.org/show_bug.cgi?id=76458
* msnconnection: Fix typoOlivier Crête2014-05-051-2/+2
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* msnconnection: Double check return value of recv()Olivier Crête2014-05-041-3/+10
| | | | Even though it has already been peeked at!
* rtpsession: Check that there is either a blueprint or a profileOlivier Crête2014-05-041-0/+6
| | | | Having neither is always invalid!
* msnconnection: Make sure token is correctly readOlivier Crête2014-05-041-1/+5
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* rtptfrc: Fix off by one errorOlivier Crête2014-05-041-1/+1
| | | | 128 is dynamic and needs checking
* rtpkeyunitmanager: Correctly check for local ssrcOlivier Crête2014-05-031-2/+3
| | | | Found by coverity
* rtpsession: Since GStreamer 1.2, the real internal SSRC is on the incoming capsOlivier Crête2014-05-031-3/+18
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* rtpsession: Also notify of SSRC change on caps changeOlivier Crête2014-05-031-0/+11
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* raw: Fix crash where the stream would try to contact its session before its ↵Olivier Crête2014-02-271-1/+2
| | | | been set
* rtpsession: Need to read the method on stop tooOlivier Crête2013-12-051-5/+4
| | | | Otherwise it used an initialized variable. Thank you clang-analyzer
* rtp-codec-nego: Actually test that the codec id is validOlivier Crête2013-11-071-1/+1
| | | | Bug found by David Binderman <dcb314@hotmail.com>
* Prefer dynamic PT 101 for telephone-event at clock rate 8000Simon McVittie2013-06-031-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | The WebRTC implementation in Google Chrome <= 26 would reject calls if there was "a telephone-event payload type less than 101"[1] and as of 2013-06-03, the Google Mail web UI with the VoIP extension seems to have a similar signalling bug. Experimenting with the web UI indicates that telephone-events with clock rate != 8000 are irrelevant, and only clock rate 8000 matters. Hopefully the same was true in WebRTC (I can't find a libjingle commit that looks likely to have fixed this). Meanwhile, many SIP implementations and at least one Jingle implementation (Freeswitch's mod_dingaling) either hard-code payload type 101 to be telephone-event, or make the payload type for telephone-event a configuration option. I can't help thinking this was not how dynamic payload types were meant to work, but interoperability is interoperability... This fixes interop when Empathy 3.8 + telepathy-gabble 0.17.4, on a system with not many codecs installed) calls the Google Mail web UI. When the same setup is called by a peer that specifies a different PT for telephone-event:8000 (the Google Mail web UI uses 126 in its outgoing calls), the peer's choice of PT takes precedence. [1] https://code.google.com/p/webrtc/issues/detail?id=1783 https://bugs.freedesktop.org/show_bug.cgi?id=65311
* fs-rtp-discover-codecs: plug memoryleakHavard Graff2013-04-081-1/+1
| | | | use g_list_delete_link to free the list as well
* Misc win32 portability fixesOlivier Crête2013-04-042-8/+5
| | | | Based on a patch by Conrad Poelman
* codec-discovery: Intersect different parts of the same caps to reduce themOlivier Crête2013-04-021-6/+34
| | | | | We do this because a caps may have the static payload in a separate structure from the encoding-name We just want both in the same structure
* rtpsession: Set error in all error casesOlivier Crête2013-04-021-2/+12
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* rtpsubstream: Don't free codec after setting it inside substreamOlivier Crête2013-03-291-8/+6
| | | | Bug discovered by Havard Graff
* rtp-stream: plug session leakHavard Graff2013-03-261-0/+3
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* fs-rtp-substream: Don't leak caps on errorOlivier Crête2013-03-261-0/+2
| | | | Based on patch from Havard Graff
* rtp-codec-discovery: Intersect instead of mergeOlivier Crête2013-03-261-6/+4
| | | | | We want the semantics of intersection, not merging, as this will produce a caps with two separate structures in some cases.
* rtp: Tune pulsesink/pulsesrc latency values furtherArun Raghavan2013-03-221-3/+2
| | | | | | This makes for lower overall values without forcing a bunch of underruns at the start which we got by having pulsesink's buffer-time as 2*latency-time.
* Use the generic marshallers instead of generating themOlivier Crête2013-02-054-48/+3
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* Update and fix the default properties for vp8encDebarshi Ray2012-10-241-4/+5
| | | | | | | | | | The property names of the vp8enc element changed in GStreamer 1.0. See the following commits from gst-plugins-good for some of the corresponding changes: - 392bd12a45b959b696365e5f25e315c2489fe025 - 9c0ff2f38174f2e4111859bd66956a77764cb515 Also, vp8enc uses target-bitrate, not bitrate.
* rtpsession: Set the discovery valve to playing before linking itOlivier Crête2012-10-041-0/+8
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