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35b09e3606
Commit Graph

1367 Commits

Author SHA1 Message Date
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
Gian-Carlo Pascutto
a466801c61 Bug 1129365 - Don't assume setPictureSize supports the same sizes as setPreviewSize. r=jesup 2015-02-05 18:24:02 +01:00
Birunthan Mohanathas
37ea92b9bf Bug 1120796 - Part 1: Prepare code for explicit bool operators. r=Waldo 2015-02-03 18:52:28 +02:00
Mike Hommey
50e6916b40 Bug 1126593 - Add a global fallible instance, so that using fallible works directly, everywhere. r=njn 2015-02-02 09:56:13 +09:00
Paul Kerr [:pkerr]
644457e17d Bug 1099318: Fix for conduit receive then send creation order issue. Now insensitive to order. r=bwc" 2015-01-29 08:52:40 -08:00
Gian-Carlo Pascutto
aad30d486a Bug 1109248: Merge with webrtc.org update (android compile/merge fixes) r=jesup 
ON A CLOSED TREE
 2015-01-29 18:34:16 -05:00
Randell Jesup
439295ff1b Bug 1109248: remove unused media/webrtc/trunk/base directory (ancient) rs=jesup 2015-01-29 18:33:36 -05:00
Randell Jesup
645d066621 Bug 1109248: Include/etc fixes for B2G from webrtc.org update rs=jesup 2015-01-29 18:33:36 -05:00
Randell Jesup
2333147169 Bug 1109248: Merge webrtc.org update with our OpenSLES changes rs=jesup 2015-01-29 18:33:36 -05:00
Gian-Carlo Pascutto
9ba529fb42 Bug 1109248: fixes for changes to webrtc Android camera fps handling r=jesup 2015-01-29 18:33:36 -05:00
Gian-Carlo Pascutto
974d6435e4 Bug 1109248: Revert removal of SetAndroidObjects calls in webrtc.org r=jesup 2015-01-29 18:33:36 -05:00
Randell Jesup
cf043c0ba4 Bug 1109248: Adapt GMP video decoder code to API changes in webrtc.org 40 r=ehugg 2015-01-29 18:33:36 -05:00
Randell Jesup
6754335535 Bug 1109248: basic adapation of new webrtc/base directory to build in mozilla rs=jesup 2015-01-29 18:33:36 -05:00
Landry Breuil
7edaac5678 Bug 1109248 - build fixes for OpenBSD r=jesup 
- check for __GLIBC__ instead of __GLIBCXX__ to include <execinfo.h>
- check for WEBRTC_BSD instead of BSD to include <stdlib.h>
 2015-01-29 18:33:36 -05:00
Randell Jesup
f004b532aa Bug 1109248: basic compile fixes for webrtc.org 40 update rs=jesup 
Mostly #ifdefing Chrome-specific code and replacing WEBRTC_TRACE with LOG_F/etc
 2015-01-29 18:33:36 -05:00
Randell Jesup
971df42891 Bug 1109248: gyp changes to adapt to webrtc.org 40 update r=ted 2015-01-29 18:33:36 -05:00
Randell Jesup
39fc4b91d6 Bug 1109248: revert removal of webrtc audio ExternalRecording interface rs=jesup 2015-01-29 18:33:36 -05:00
Randell Jesup
6b85bfac08 Bug 1109248: Revert webrtc upstream Issue 18399004 which removed APIs we're using rs=jesup 
https://webrtc-codereview.appspot.com/18399004
 2015-01-29 18:33:36 -05:00
Randell Jesup
d38e4b8d98 Bug 1109248: Rollup of changes previously applied to media/webrtc/trunk/webrtc rs=jesup 2015-01-29 18:33:36 -05:00
Randell Jesup
ab0973c6bc Bug 1109248: Webrtc updated to branch 40 7864; pull made Wed Dec 10 12:23:33 EST 2014 rs=jesup 2015-01-29 18:33:35 -05:00
Byron Campen [:bwc]
f5de114332 Bug 1095218 - Part 2: Multistream support. r=mt 2014-12-10 11:17:09 -08:00
Byron Campen [:bwc]
9eb5462cd8 Bug 1095218 - Part 1: msid support. r=mt 2014-12-01 21:19:57 -08:00
Byron Campen [:bwc]
b0a72b6229 Bug 1126036: Queue runnables for starting gathering and checking in PCMedia until the proxy lookup is complete. r=mt 2015-01-26 15:24:37 -08:00
Ethan Hugg
0203e72f33 Bug 1125047 - GMP should catch decoder failures r=jesup 2015-01-26 15:00:06 -08:00
6835d07c00 Bug 949703 - Part 1: Use HTTP proxy for WebRTC TURN/TCP r=ekr 2015-01-21 16:26:00 -08:00
Matthew Gregan
52d8d7d6ac Bug 1124542 - WebrtcGmpVideoDecoder shouldn't crash when GMP completion callbacks are received. r=rjesup 2015-01-21 20:26:00 +13:00
Matthew Gregan
a33164d5da Bug 1124021 - Fix dangerous UniquePtr usage pattern in GMP. r=cpearce 2015-01-20 18:39:00 +13:00
Wes Kocher
1256697bfb Backed out 2 changesets (bug 949703) for mochitest-e10s-3 orange 
Backed out changeset 6f5a7adab265 (bug 949703)
Backed out changeset 7933aeabf6bd (bug 949703)
 2015-01-21 17:15:02 -08:00
Byron Campen [:bwc]
2f47f1c8f7 Bug 949703 - Part 2: Consolidate the two copies of DummySocket we have floating around. r=drno 2014-12-19 11:11:02 -08:00
543ee9a01e Bug 949703 - Part 1: Use HTTP proxy for WebRTC TURN/TCP r=ekr 2015-01-18 11:51:00 -08:00