Byron Campen [:bwc]
|
d19ee72543
|
Bug 1133051: Clean up SctpFlows on STS r=mt
--HG--
extra : rebase_source : cbc09ce944e4580e3e190766767612b98a8ced61
|
2015-02-13 13:32:01 -08:00 |
|
Randell Jesup
|
fef42c5c7a
|
Bug 1137472: test vp9 sdp in sdp_unittests r=bwc
|
2015-03-03 23:46:16 -05:00 |
|
Randell Jesup
|
00a74cff8f
|
Bug 1137472: Basic VP9 signaling/pipeline/conduit support r=bwc
|
2015-03-03 01:31:33 -05:00 |
|
Randell Jesup
|
c9b6a04ac4
|
Bug 1137474: Fix depacketization of "Generic" encoded RTP video r=pkerr
|
2015-03-03 01:31:33 -05:00 |
|
Randell Jesup
|
f91175ae7f
|
Bug 1137474: Basic vp9 support added to upstream (using 'generic' packetization) r=pkerr
|
2015-03-03 01:31:33 -05:00 |
|
Andreas Pehrson
|
9b19093204
|
Bug 1129263 - Part 6. Remove DOMMediaStream::TrackTypeHints. r=roc,jesup
|
2015-02-09 15:23:34 +08:00 |
|
Andreas Pehrson
|
68db4947b3
|
Bug 1129263 - Part 5. Add intial remote PeerConnection tracks atomically to SourceMediaStream. r=jesup
|
2015-02-11 16:21:11 +08:00 |
|
Chris Peterson
|
e4a0d35e15
|
Bug 1136004 - Fix -Wthread-safety-analysis warning in webrtc. r=jesup
|
2015-03-02 19:51:29 -08:00 |
|
Byron Campen [:bwc]
|
c1cdd99c64
|
Bug 1133866: Some refactoring and simplification in JsepSessionImpl. r=mt
--HG--
extra : rebase_source : 64e445b182c7b1fad514d354cc6bf1f4abfecd7f
|
2015-02-25 08:36:01 -08:00 |
|
Ethan Hugg
|
b7a81f04b3
|
Bug 1137508 Change H264 maxPayloadSize to 0 for Mode 1 r=jesup
|
2015-02-26 15:29:36 -08:00 |
|
Gian-Carlo Pascutto
|
1482147cb9
|
Bug 1123012 - Just return a NULL ptr instead of casting NULL. r=jesup
|
2015-02-25 08:31:11 +01:00 |
|
Byron Campen [:bwc]
|
1c767dd0c0
|
Bug 1135902: Set stream id on fake media streams. r=drno
--HG--
extra : rebase_source : edf3e19236c05d6040ae2320d7f385105a30efe2
extra : amend_source : 79efc1ed9537b5710516e9883dc91f45c344847e
|
2015-02-23 15:19:17 -08:00 |
|
Ehsan Akhgari
|
5bff0c0b03
|
Bug 1135753 - Mark some overridden virtual functions in WebRTC as MOZ_OVERRIDE; r=mt
|
2015-02-24 09:51:57 -05:00 |
|
Randell Jesup
|
05c71da4ba
|
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
|
71cad0a045
|
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
|
1d16a313f3
|
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]
|
b25a2bb677
|
Bug 1089798 - MediaStream ID tests. r=bwc
|
2015-02-17 22:54:00 -05:00 |
|
Nils Ohlmeier [:drno]
|
629772a073
|
Bug 1089798 - Implemenation of MediaStream IDs. r=bwc
|
2015-02-19 12:59:00 -05:00 |
|
Steve Singer
|
4f3952c197
|
Bug 1130223 - Add an implementation to the big endian conditional. r=jesup
|
2015-02-15 09:36:00 +01:00 |
|
Byron Campen [:bwc]
|
794f966f80
|
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]
|
248d6e6d79
|
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]
|
1f815978b4
|
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
|
cc4a1f03e4
|
Bug 1131960 - Check for NEON capability before using NEON code. r=derf
CLOSED TREE
|
2015-02-13 05:13:00 -05:00 |
|
Randell Jesup
|
40f518f5bf
|
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
|
e29d8e1845
|
Bug 1127642 WebRTC support for H.264 max_mbps r=jesup
|
2015-02-12 11:14:57 +08:00 |
|
Randell Jesup
|
687807441c
|
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
|
c7a5446fd2
|
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
|
554ea67ff5
|
Bug 1131340 - Avoid template aliasing since GCC 4.6 lacks support. r=cpearce
|
2015-02-10 14:27:36 +13:00 |
|
Nicholas Nethercote
|
242708cf72
|
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
|
a85bba5efb
|
Bug 1130290 - Remove PeerConnectionImpl::CreateFakeMediaStream. r=jesup
--HG--
extra : rebase_source : c5fe9a894178e600c48ce22e45b9c124c76cf712
|
2015-02-05 23:56:00 +01:00 |
|
Andrew McCreight
|
d3826daa16
|
Back out Bug 1127201 (part 2) for various problems.
|
2015-02-06 15:04:32 -08:00 |
|
Nicholas Nethercote
|
d34f0301b8
|
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
|
9134477c28
|
Bug 1129921 - Account for stopCapture possibly being called twice. r=jesup
|
2015-02-05 18:24:02 +01:00 |
|
Gian-Carlo Pascutto
|
58eb5e24e0
|
Bug 1129858 - Get the local preview surface (line dropped during merge). r=jesup
|
2015-02-05 18:24:02 +01:00 |
|
Gian-Carlo Pascutto
|
e8ec6fb3c4
|
Bug 1129365 - Don't assume setPictureSize supports the same sizes as setPreviewSize. r=jesup
|
2015-02-05 18:24:02 +01:00 |
|
Birunthan Mohanathas
|
847dbb9825
|
Bug 1120796 - Part 1: Prepare code for explicit bool operators. r=Waldo
|
2015-02-03 18:52:28 +02:00 |
|
Mike Hommey
|
a35dbaeebf
|
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]
|
0dd312ac8c
|
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
|
adfc170313
|
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
|
f82b381b47
|
Bug 1109248: remove unused media/webrtc/trunk/base directory (ancient) rs=jesup
|
2015-01-29 18:33:36 -05:00 |
|
Randell Jesup
|
96d17ba850
|
Bug 1109248: Include/etc fixes for B2G from webrtc.org update rs=jesup
|
2015-01-29 18:33:36 -05:00 |
|
Randell Jesup
|
a333fa4da0
|
Bug 1109248: Merge webrtc.org update with our OpenSLES changes rs=jesup
|
2015-01-29 18:33:36 -05:00 |
|
Gian-Carlo Pascutto
|
2bd1c1b6a1
|
Bug 1109248: fixes for changes to webrtc Android camera fps handling r=jesup
|
2015-01-29 18:33:36 -05:00 |
|
Gian-Carlo Pascutto
|
715dfa95f8
|
Bug 1109248: Revert removal of SetAndroidObjects calls in webrtc.org r=jesup
|
2015-01-29 18:33:36 -05:00 |
|
Randell Jesup
|
37194f082b
|
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
|
d17d6d6c85
|
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
|
5fa8bc8fb9
|
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
|
47e542881b
|
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
|
ead017e967
|
Bug 1109248: gyp changes to adapt to webrtc.org 40 update r=ted
|
2015-01-29 18:33:36 -05:00 |
|
Randell Jesup
|
100c8393ed
|
Bug 1109248: revert removal of webrtc audio ExternalRecording interface rs=jesup
|
2015-01-29 18:33:36 -05:00 |
|