Commit Graph

1396 Commits

Author SHA1 Message Date
Byron Campen [:bwc]
00ec4855fd Bug 1136252 - Part 2: Wait for less RTP in signaling_unittests. r=mt 2015-03-06 15:16:38 -08:00
Karina Li
20dd043a33 Bug 1138320 - Set screensharing mode or video mode for WebRTC video sources r=jesup 2015-03-12 14:14:29 +08:00
Sotaro Ikeda
705b478dfa Bug 1137515 part 2 - Change to media r=jesup 2015-03-11 12:32:38 -07:00
Jan-Ivar Bruaroey
28242b71ba Bug 1136871 - cleanup RtpSenders accounting to not rely on streams r=mt 2015-03-11 12:24:38 -04:00
Byron Campen [:bwc]
197bed0e65 Bug 1136871 - Part 2: Clear up some inconsistencies with ReplaceTrack r=jib,smaug 2015-03-11 12:08:21 -04:00
Ryan VanderMeulen
0c9cbc27b9 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
f119b09b55 Bug 1137515 part 2 - Change to media r=jesup 2015-03-11 07:18:23 -07:00
Byron Campen [:bwc]
cae69e38d3 Bug 1140635: Remove |magic_num| fields from sipcc. 2015-03-06 15:58:53 -08:00
Byron Campen [:bwc]
63b15e6e03 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
665d625c2f Merge b2g-inbound to m-c a=merge CLOSED TREE 2015-03-10 15:44:53 -07:00
Carsten "Tomcat" Book
df94ebd0ae Merge mozilla-central to b2g-inbound 2015-03-10 14:07:36 +01:00
Thomas Zimmermann
5b4403e770 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]
3a4900388d 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
07fa46f676 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
410f8f621f Bug 1140677 - Add RTPFragmentationHeader handling on gonk r=jesup 2015-03-09 18:36:23 -07:00
Ethan Hugg
3ac0d1313e 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]
e5c962d6d8 Bug 1133051: Clean up SctpFlows on STS r=mt 2015-02-13 13:32:01 -08:00
Randell Jesup
287d4b458d Bug 1137472: test vp9 sdp in sdp_unittests r=bwc 2015-03-03 23:46:16 -05:00
Randell Jesup
ca62a50ed0 Bug 1137472: Basic VP9 signaling/pipeline/conduit support r=bwc 2015-03-03 01:31:33 -05:00
Randell Jesup
264940a91d Bug 1137474: Fix depacketization of "Generic" encoded RTP video r=pkerr 2015-03-03 01:31:33 -05:00
Randell Jesup
fcf90a49d8 Bug 1137474: Basic vp9 support added to upstream (using 'generic' packetization) r=pkerr 2015-03-03 01:31:33 -05:00
Andreas Pehrson
11057e57a1 Bug 1129263 - Part 6. Remove DOMMediaStream::TrackTypeHints. r=roc,jesup 2015-02-09 15:23:34 +08:00
Andreas Pehrson
d3e1679cb5 Bug 1129263 - Part 5. Add intial remote PeerConnection tracks atomically to SourceMediaStream. r=jesup 2015-02-11 16:21:11 +08:00
Chris Peterson
92dd727714 Bug 1136004 - Fix -Wthread-safety-analysis warning in webrtc. r=jesup 2015-03-02 19:51:29 -08:00
Byron Campen [:bwc]
e6d79f9627 Bug 1133866: Some refactoring and simplification in JsepSessionImpl. r=mt 2015-02-25 08:36:01 -08:00
Ethan Hugg
c740f6dda1 Bug 1137508 Change H264 maxPayloadSize to 0 for Mode 1 r=jesup 2015-02-26 15:29:36 -08:00
Gian-Carlo Pascutto
13f13d3a23 Bug 1123012 - Just return a NULL ptr instead of casting NULL. r=jesup 2015-02-25 08:31:11 +01:00
Byron Campen [:bwc]
def936e25d Bug 1135902: Set stream id on fake media streams. r=drno 2015-02-23 15:19:17 -08:00
Ehsan Akhgari
fce72f0392 Bug 1135753 - Mark some overridden virtual functions in WebRTC as MOZ_OVERRIDE; r=mt 2015-02-24 09:51:57 -05:00
Randell Jesup
35b09e3606 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
a5866d791e 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
8bdecb1c77 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]
e701d38674 Bug 1089798 - MediaStream ID tests. r=bwc 2015-02-17 22:54:00 -05:00
Nils Ohlmeier [:drno]
a73c305eae Bug 1089798 - Implemenation of MediaStream IDs. r=bwc 2015-02-19 12:59:00 -05:00
Steve Singer
253bfe2c00 Bug 1130223 - Add an implementation to the big endian conditional. r=jesup 2015-02-15 09:36:00 +01:00
Byron Campen [:bwc]
211a2eeadf Bug 1130534: Use a single bidirectional media conduit that MediaPipelines can attach/detach at will. r=jesup 2015-02-10 10:11:24 -08:00
Byron Campen [:bwc]
75c217fe3a Bug 1017888 - Part 2: Testing for renegotiation. r=mt, r=drno 2015-02-10 10:17:03 -08:00
Byron Campen [:bwc]
f57e11b2da Bug 1017888 - Part 1: Renegotiation support. r=mt, r=smaug 2014-12-10 15:53:54 -08:00
Gian-Carlo Pascutto
1d91676b66 Bug 1131960 - Check for NEON capability before using NEON code. r=derf
CLOSED TREE
2015-02-13 05:13:00 -05:00
Randell Jesup
66bc848b22 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
4daf73521c Bug 1127642 WebRTC support for H.264 max_mbps r=jesup 2015-02-12 11:14:57 +08:00
Randell Jesup
461b8bc71e 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
9f0518d6d2 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
1a7591bf44 Bug 1131340 - Avoid template aliasing since GCC 4.6 lacks support. r=cpearce 2015-02-10 14:27:36 +13:00
Nicholas Nethercote
ee41df7dc2 Bug 1127201 (attempt 2, part 1) - Replace most NS_ABORT_IF_FALSE calls with MOZ_ASSERT. r=Waldo. 2015-02-09 14:34:50 -08:00
Andreas Pehrson
a194bc9f62 Bug 1130290 - Remove PeerConnectionImpl::CreateFakeMediaStream. r=jesup 2015-02-05 23:56:00 +01:00
Andrew McCreight
1ee96e7527 Back out Bug 1127201 (part 2) for various problems. 2015-02-06 15:04:32 -08:00
Nicholas Nethercote
0a02b5d31c Bug 1127201 (part 2) - Convert all NS_ABORT_IF_FALSE calls to MOZ_ASSERT. r=Waldo. 2015-02-04 20:05:36 -08:00
Gian-Carlo Pascutto
f2529ddaab Bug 1129921 - Account for stopCapture possibly being called twice. r=jesup 2015-02-05 18:24:02 +01:00
Gian-Carlo Pascutto
25d94694e9 Bug 1129858 - Get the local preview surface (line dropped during merge). r=jesup 2015-02-05 18:24:02 +01:00