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4cb7e10356
Commit Graph

1378 Commits

Author SHA1 Message Date
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
Andreas Pehrson
5112c3404f Bug 1130290 - Remove PeerConnectionImpl::CreateFakeMediaStream. r=jesup 
--HG--
extra : rebase_source : c5fe9a894178e600c48ce22e45b9c124c76cf712
 2015-02-05 23:56:00 +01:00
Andrew McCreight
8413cc973c Back out Bug 1127201 (part 2) for various problems. 2015-02-06 15:04:32 -08:00
Nicholas Nethercote
3629781b69 Bug 1127201 (part 2) - Convert all NS_ABORT_IF_FALSE calls to MOZ_ASSERT. r=Waldo. 
--HG--
extra : rebase_source : 99182e70335d2b5ff95f8c528ae992d37294be3a
 2015-02-04 20:05:36 -08:00
Gian-Carlo Pascutto
54320124a9 Bug 1129921 - Account for stopCapture possibly being called twice. r=jesup 2015-02-05 18:24:02 +01:00
Gian-Carlo Pascutto
3b48ccfc4f Bug 1129858 - Get the local preview surface (line dropped during merge). r=jesup 2015-02-05 18:24:02 +01:00
Gian-Carlo Pascutto
66465bbb49 Bug 1129365 - Don't assume setPictureSize supports the same sizes as setPreviewSize. r=jesup 2015-02-05 18:24:02 +01:00
Birunthan Mohanathas
318898d688 Bug 1120796 - Part 1: Prepare code for explicit bool operators. r=Waldo 2015-02-03 18:52:28 +02:00
Mike Hommey
824818ee98 Bug 1126593 - Add a global fallible instance, so that using fallible works directly, everywhere. r=njn 
--HG--
rename : memory/mozalloc/fallible.h => memory/fallible/fallible.h
 2015-02-02 09:56:13 +09:00
Paul Kerr [:pkerr]
9f7b135063 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
5ffc8692cf 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
10dd4a6dac Bug 1109248: remove unused media/webrtc/trunk/base directory (ancient) rs=jesup 2015-01-29 18:33:36 -05:00
Randell Jesup
9fecbd3568 Bug 1109248: Include/etc fixes for B2G from webrtc.org update rs=jesup 2015-01-29 18:33:36 -05:00
Randell Jesup
44dcdd8ab9 Bug 1109248: Merge webrtc.org update with our OpenSLES changes rs=jesup 2015-01-29 18:33:36 -05:00
Gian-Carlo Pascutto
70fd7cc0b0 Bug 1109248: fixes for changes to webrtc Android camera fps handling r=jesup 2015-01-29 18:33:36 -05:00
Gian-Carlo Pascutto
d78166f848 Bug 1109248: Revert removal of SetAndroidObjects calls in webrtc.org r=jesup 2015-01-29 18:33:36 -05:00
Randell Jesup
b486978770 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
782ce26312 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
282a9799e6 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
205d235ac7 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
855e7f4116 Bug 1109248: gyp changes to adapt to webrtc.org 40 update r=ted 2015-01-29 18:33:36 -05:00
Randell Jesup
05d3f1973f Bug 1109248: revert removal of webrtc audio ExternalRecording interface rs=jesup 2015-01-29 18:33:36 -05:00
Randell Jesup
00e69d533a 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
c2f913d156 Bug 1109248: Rollup of changes previously applied to media/webrtc/trunk/webrtc rs=jesup 2015-01-29 18:33:36 -05:00