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abf4565524
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1401 Commits

Author SHA1 Message Date
Byron Campen [:bwc]
abf4565524 Bug 1141652: Simplify RTCP bundle filtering to only filter for receive pipelines, based only on the SR SSRC. r=jesup 2015-03-12 09:08:13 -07:00
Byron Campen [:bwc]
e1bf9ab649 Bug 1136252 - Part 6: Extend timeouts for RTP/RTCP, until bug 1137948 can be fixed. r=mt 2015-03-06 15:16:39 -08:00
Byron Campen [:bwc]
39d440e134 Bug 1136252 - Part 5: Fix bug where candidates could be trickled before setRemote during renegotiation. r=mt 2015-03-06 15:16:38 -08:00
Byron Campen [:bwc]
0e15eb28c4 Bug 1136252 - Part 4: Remove/consolidate code in signaling_unittests. Includes removing most SDP checks, since that belongs in jsep_session_unittest. r=mt 2015-03-06 15:16:38 -08:00
Byron Campen [:bwc]
d085984c4d Bug 1136252 - Part 3: Remove some unnecessary sleeps in signaling_unittests. r=mt 2015-03-06 15:16:38 -08:00
Byron Campen [:bwc]
9077106c43 Bug 1136252 - Part 2: Wait for less RTP in signaling_unittests. r=mt 2015-03-06 15:16:38 -08:00
Karina Li
b6128d188d Bug 1138320 - Set screensharing mode or video mode for WebRTC video sources r=jesup 2015-03-12 14:14:29 +08:00
Sotaro Ikeda
4990e78217 Bug 1137515 part 2 - Change to media r=jesup 2015-03-11 12:32:38 -07:00
Jan-Ivar Bruaroey
d56642e463 Bug 1136871 - cleanup RtpSenders accounting to not rely on streams r=mt 2015-03-11 12:24:38 -04:00
Byron Campen [:bwc]
1fc68b315e Bug 1136871 - Part 2: Clear up some inconsistencies with ReplaceTrack r=jib,smaug 2015-03-11 12:08:21 -04:00
Ryan VanderMeulen
3b328f07d7 Backed out changesets cd5ec762afa1 and fad66e8fe874 (bug 1137515) for Nexus 5-L bustage. 
CLOSED TREE
 2015-03-11 12:35:08 -04:00
Sotaro Ikeda
f1036a12da Bug 1137515 part 2 - Change to media r=jesup 2015-03-11 07:18:23 -07:00
Byron Campen [:bwc]
0e8278219d Bug 1140635: Remove |magic_num| fields from sipcc. 2015-03-06 15:58:53 -08:00
Byron Campen [:bwc]
2a9d3f8962 Bug 1140089: Call SetPullEnabled on all streams in PCMedia when offer/answer concludes. r=jesup 2015-03-06 14:37:11 -08:00
Wes Kocher
f303b4d81c Merge b2g-inbound to m-c a=merge CLOSED TREE 2015-03-10 15:44:53 -07:00
Carsten "Tomcat" Book
b3a1935b89 Merge mozilla-central to b2g-inbound 2015-03-10 14:07:36 +01:00
Thomas Zimmermann
f87dc13636 Bug 1137151: Remove ref-counting from |OMXVideoEncoder| r=sotaro 
Reference counting in |OMXVideoEncoder| is used inconsistently any actually
not necessary. This patch removed the code. Users are converted to auto
pointers.
 2015-03-10 13:44:01 +01:00
Byron Campen [:bwc]
555ae0c048 Bug 1140637: Add jsep_session_unittest to testing/cppunittest.ini, and unbust it. r=jesup 2015-03-09 14:45:46 -07:00
Cesar Guirao
b76a0d118f Bug 1139132: Fix Chroma offset on WebRTC remote video when width is not even r=jesup 
Fixed chroma plane offset calculation when frame width/height is not even
 2015-03-03 21:06:00 +01:00
Sotaro Ikeda
48a4de554b Bug 1140677 - Add RTPFragmentationHeader handling on gonk r=jesup 2015-03-09 18:36:23 -07:00
Ethan Hugg
1c787bc0dc Bug 1140648 - WebRTC check codec config max frame rate is set before using r=jesup 2015-03-06 19:05:11 -08:00
Byron Campen [:bwc]
03d3717cb8 Bug 1133051: Clean up SctpFlows on STS r=mt 2015-02-13 13:32:01 -08:00
Randell Jesup
467a185a7a Bug 1137472: test vp9 sdp in sdp_unittests r=bwc 2015-03-03 23:46:16 -05:00
Randell Jesup
4cb7e10356 Bug 1137472: Basic VP9 signaling/pipeline/conduit support r=bwc 2015-03-03 01:31:33 -05:00
Randell Jesup
c26ab8cd38 Bug 1137474: Fix depacketization of "Generic" encoded RTP video r=pkerr 2015-03-03 01:31:33 -05:00
Randell Jesup
52aaf48f23 Bug 1137474: Basic vp9 support added to upstream (using 'generic' packetization) r=pkerr 2015-03-03 01:31:33 -05:00
Andreas Pehrson
d7ef0ecef9 Bug 1129263 - Part 6. Remove DOMMediaStream::TrackTypeHints. r=roc,jesup 2015-02-09 15:23:34 +08:00
Andreas Pehrson
ca37ef041b Bug 1129263 - Part 5. Add intial remote PeerConnection tracks atomically to SourceMediaStream. r=jesup 2015-02-11 16:21:11 +08:00
Chris Peterson
4c4c0714a7 Bug 1136004 - Fix -Wthread-safety-analysis warning in webrtc. r=jesup 2015-03-02 19:51:29 -08:00
Byron Campen [:bwc]
68d2da98bc Bug 1133866: Some refactoring and simplification in JsepSessionImpl. r=mt 2015-02-25 08:36:01 -08:00
Ethan Hugg
4a2e45feb2 Bug 1137508 Change H264 maxPayloadSize to 0 for Mode 1 r=jesup 2015-02-26 15:29:36 -08:00
Gian-Carlo Pascutto
8275a74b81 Bug 1123012 - Just return a NULL ptr instead of casting NULL. r=jesup 2015-02-25 08:31:11 +01:00
Byron Campen [:bwc]
5755021607 Bug 1135902: Set stream id on fake media streams. r=drno 2015-02-23 15:19:17 -08:00
Ehsan Akhgari
245c448238 Bug 1135753 - Mark some overridden virtual functions in WebRTC as MOZ_OVERRIDE; r=mt 2015-02-24 09:51:57 -05:00
Randell Jesup
88e723928b Bug 1136004: fix TSAN warning in webrtc when RED isn't enabled r=cpeterson 2015-02-24 02:08:04 -05:00
Gian-Carlo Pascutto
63c2a16c8e Bug 1134991 - Failure to set up voice communication mode in OpenSLES should not be fatal. r=jesup 2015-02-20 19:13:13 +01:00
Randell Jesup
99d02c92e9 Bug 1128116: Fix decoding H264 in webrtc where SPS & PPS aren't in a STAP-A packet r=ehugg 
FF 37 and before didn't encode SPS/PPS into a STAP-A packet, and the
webrtc.org in branch 40 code doesn't handle that (common) case.
 2015-02-22 19:10:59 -05:00
Nils Ohlmeier [:drno]
636ec3100e Bug 1089798 - MediaStream ID tests. r=bwc 2015-02-17 22:54:00 -05:00
Nils Ohlmeier [:drno]
3201a1b72b Bug 1089798 - Implemenation of MediaStream IDs. r=bwc 2015-02-19 12:59:00 -05:00
Steve Singer
05432acaec Bug 1130223 - Add an implementation to the big endian conditional. r=jesup 2015-02-15 09:36:00 +01:00
Byron Campen [:bwc]
a78f5b1da1 Bug 1130534: Use a single bidirectional media conduit that MediaPipelines can attach/detach at will. r=jesup 
--HG--
extra : rebase_source : 202a83e513d88bc14f1be2c5b438998461ff4a50
 2015-02-10 10:11:24 -08:00
Byron Campen [:bwc]
3a680bf820 Bug 1017888 - Part 2: Testing for renegotiation. r=mt, r=drno 
--HG--
extra : rebase_source : 7434ef35ea6294966220f20431941e0790c4221c
 2015-02-10 10:17:03 -08:00
Byron Campen [:bwc]
eee4f8cd6d Bug 1017888 - Part 1: Renegotiation support. r=mt, r=smaug 
--HG--
extra : rebase_source : df1c2962ee88f75c6ad676b9cd79978a87dafb65
extra : amend_source : c938904331323ff3565624795ac76d82502f43fb
 2014-12-10 15:53:54 -08:00
Gian-Carlo Pascutto
ce9164c343 Bug 1131960 - Check for NEON capability before using NEON code. r=derf 
CLOSED TREE
 2015-02-13 05:13:00 -05:00
Randell Jesup
f01e29d351 Bug 1108248: Swap CreateTimerQueueTimer() for timerSetEvent() in webrtc win32 code r=dmajor 
Avoids limits on the number of realtime (timerSetEvent()) timers
 2015-02-06 17:24:50 -05:00
Karina Li
956c026bed Bug 1127642 WebRTC support for H.264 max_mbps r=jesup 2015-02-12 11:14:57 +08:00
Randell Jesup
fafce09a24 Bug 1132193: Re-enable AEC debug logging in getUserMedia r=pkerr 
Temporarily disabled by landing for upstream webrtc branch 40.  Also saves
as .wav format now
 2015-02-12 07:46:59 -05:00
Randell Jesup
0c7c2dd363 Bug 1124175: Remove limits on odd webrtc downsample sizes due to load/bitrate r=pkerr 
Also convert assert to limits on max size
 2015-02-11 17:29:01 -05:00
Matthew Gregan
100e19229b Bug 1131340 - Avoid template aliasing since GCC 4.6 lacks support. r=cpearce 2015-02-10 14:27:36 +13:00
Nicholas Nethercote
6ee3666899 Bug 1127201 (attempt 2, part 1) - Replace most NS_ABORT_IF_FALSE calls with MOZ_ASSERT. r=Waldo. 
--HG--
extra : rebase_source : 488e401ff87e31a2074c4108c4df0572d9536667
 2015-02-09 14:34:50 -08:00