6556e8e870
Bug 821066 - Remove dead remotedesc/localdesc messages and events from sipcc. r=abr
2013-02-04 16:18:26 -05:00
39786b0d47
Bug 837421: Webrtc: Ignore second call to ConnectDataConnection r=derf
2013-02-03 00:29:04 -05:00
6e1c0719df
Merge mozilla-central into mozilla-inbound
2013-02-02 13:53:46 -05:00
7c163ea485
Bug 788185: add a/v sync to Audio/Video Conduits r=derf
2013-02-02 08:46:54 -05:00
f8ab8f1ffe
Bug 823056: Move locks to protect linked_ptr<> instances r=ehugg
2013-02-01 16:19:24 -06:00
4e5c2a4b8a
Bug 830707. Part 3: Don't constrain AudioSegment to a fixed number of channels. r=jesup
...
--HG--
extra : rebase_source : feacede00821b6673ce04c886a9c3727a4989404
2013-01-21 09:44:44 +13:00
a681906644
Bug 834383: Ensure PeerConnectionImpl destructor doesn't use globals after they're gone r=jesup,bsmith
2013-01-30 15:31:22 -06:00
a3d39b9bfa
Bug 834100 - Null deref if you call addIceCandidate on an RTCPeerConnection before setting localDesc [@ fsmdef_ev_addcandidate]. r=abr
2013-01-31 15:43:03 -05:00
c66b40cd52
Backed out changeset 70872c020944 (bug 834383) speculatively to see if it fixes the random timeouts in test_peerConnection_basicAudio.html and friends.
...
CLOSED TREE
2013-01-31 20:24:18 -05:00
373b364c09
Backed out changeset b1e8d7154488 (bug 834100) for Fedora64 M3 orange.
2013-01-31 17:14:13 -05:00
f58619e166
Bug 834100 - Null deref if you call addIceCandidate on an RTCPeerConnection before setting localDesc [@ fsmdef_ev_addcandidate]. r=abr
2013-01-31 15:43:03 -05:00
1ce94cf955
Bug 834383: Ensure PeerConnectionImpl destructor doesn't use globals after they're gone r=jesup,bsmith
2013-01-30 15:31:22 -06:00
500fe870eb
Bug 818670: Ensure PipelineListener doesn't release conduit off main thread r=derf
2013-01-30 21:26:45 -05:00
a072473354
Bug 818670: merge AudioConduits to allow AEC to work r=ekr
2013-01-29 11:57:44 -05:00
83c2fd91ec
Bug 818670: Enable AEC in PeerConnection, AGC/NoiseSuppression in gUM (w/bustage fix) r=derf
2013-01-29 11:55:09 -05:00
dff32419e8
Bug 812278 followup: Remove a few more unnecessary semicolons after MOZ_MTLOG_MODULE(), for consistency. (no review)
2013-01-30 13:13:36 -08:00
58ea6b383c
Backout 40f09f7bc670 & fc262e3c635f (bug 818670) for frequent fedora64 mochitest-3 leaks on a CLOSED TREE
2013-01-30 10:32:11 +00:00
929d986e52
Bug 834768: Fix signed/unsigned comparison build warnings in signaling_unittests.cpp. r=jesup
2013-01-29 20:03:42 -08:00
505ded4fda
Bug 818670: merge AudioConduits to allow AEC to work r=ekr
2013-01-29 11:57:44 -05:00
5bfe28ca5c
Bug 818670: Enable AEC in PeerConnection, AGC/NoiseSuppression in gUM (w/bustage fix) r=derf
2013-01-29 11:55:09 -05:00
367cd59d9d
Backout df75a87cce60 & 19e164f7d88d (bug 818670) for build bustage on a CLOSED TREE
2013-01-29 17:28:30 +00:00
38a431722d
Bug 818670: merge AudioConduits to allow AEC to work r=ekr
2013-01-29 11:57:44 -05:00
abaf7857dc
Bug 818670: Enable AEC in PeerConnection, AGC/NoiseSuppression in gUM r=derf
2013-01-29 11:55:09 -05:00
c6638a677b
Bug 830099. Flag WebIDL dictionaries and callbacks with some information indicating whether we need main-thread and worker codegen for them and then use that information to skip unneccessary codegen. r=peterv
2013-01-28 23:30:17 -05:00
426c47cdf0
Back out c5238879470f and b3cabb259af7 (bug 830099) for bustage
2013-01-28 22:36:59 -08:00
719c7bf030
Bug 830099. Flag WebIDL dictionaries and callbacks with some information indicating whether we need main-thread and worker codegen for them and then use that information to skip unneccessary codegen. r=peterv
2013-01-28 23:30:17 -05:00
aba52e62fb
Bug 835290 MediaPipeline - replace attempted += of ints with PR_snprintfs r=jesup
2013-01-28 12:26:32 -08:00
6e6d470f56
Bug 816780 - Merge all incoming m-lines into one MediaStream. r=jesup,abr
2013-01-24 08:34:18 -08:00
00a0bba88a
Backed out changeset 9b803c2821b9 (bug 818843) due to B2G device build breakage on Mac. rs=jesup
...
--HG--
extra : rebase_source : 981d5e9504c69eb76112fca35fd0a36110593659
2013-01-25 17:14:05 -05:00
0995673625
Bug 834463: Corrected RTCConfiguration format. r=jst
2013-01-24 17:58:29 -05:00
4da08a17f1
Bug 825703: Stun configuration from JS for PeerConnections (IP only) r=bz,jesup
2013-01-23 14:21:25 -05:00
a9a1f3a5ed
Bug 786236: Per-context configurable STUN servers. r=abr
2013-01-05 11:40:34 -08:00
7bec764b3a
Bug 832683: Match SRTP policy values to enable NACK mode in webrtc r=ekr
2013-01-23 16:41:35 -05:00
28db8f365c
Bug 818843 - Media changes. r=rjesup
2013-01-23 18:08:16 -05:00
932560ee43
Bug 806825 - Fix possible data race of PeerConnectionCtx::mSipccState r=jesup
2013-01-22 11:25:04 -08:00
9bf70cc915
Bug 829757 - Clean up a=candidate stripping. r=abr
2013-01-19 15:18:28 -08:00
dff20ff30d
Follow-up for bug 833097, bug 833101 and bug 833118 - Only disable PGO on the modules in question on Windows, a=me
...
--HG--
extra : rebase_source : dae7ac9db9e0b85a39cf7d647b46f9da48398dc2
2013-01-21 18:53:05 -05:00
295bf1fa7a
Bug 833118 - Disable PGO on webrtc; a=me
2013-01-21 18:02:12 -05:00
ca3161f4d1
Bug 832567: Adjust minimum default bitrate to 200Kbps until we support lower resolutions rs=me
2013-01-21 03:49:57 -05:00
d8bbecd100
Bug 832567: transmitting-only channels weren't accepting RTCP; turn off NACK mode and block bad outgoing RTCP from transmit-only channels r=ekr
2013-01-21 03:49:55 -05:00
aac86acc69
Bug 394311 - Stop building with -pedantic. r=ted,dbaron.
...
--HG--
rename : content/svg/content/nsISVGPoint.cpp => content/svg/content/src/DOMSVGPoint.cpp
rename : docshell/test/browser/browser_bug234628-9.js => docshell/test/browser/browser_bug134911.js
rename : layout/reftests/w3c-css/submitted/values3/calc-background-image-gradient-1-ref.html => layout/reftests/css-calc/background-image-gradient-1-ref.html
rename : layout/reftests/w3c-css/submitted/values3/calc-background-image-gradient-1.html => layout/reftests/css-calc/background-image-gradient-1.html
rename : layout/reftests/w3c-css/submitted/values3/reftest.list => layout/reftests/css-calc/reftest.list
rename : layout/reftests/text/auto-hyphenation-10-ref.html => layout/reftests/text/auto-hyphenation-10.html
rename : layout/reftests/text/auto-hyphenation-8-ref.html => layout/reftests/text/auto-hyphenation-8.html
rename : layout/reftests/text/auto-hyphenation-9-ref.html => layout/reftests/text/auto-hyphenation-9.html
extra : rebase_source : 012df725d55b031ccc03d9bfcf785056d95a2ebe
2013-01-20 14:12:42 -08:00
615b59944e
Bug 831764. Reverse DTLS client/server selection to more closely match RFC 5763. r=jesup
2013-01-17 07:54:53 -08:00
ddea6fdd5f
Bug 829624 - Always use 80-bit MAC with SRTCP. r=jesup
2013-01-18 13:03:17 -08:00
e940f1fcfd
Bug 831831: Don't ignore incoming RTCP; don't make webrtc code think no bytes were sent r=ekr
2013-01-18 10:45:53 -05:00
c26dd6c6e1
Bug 824919: Weaken key references to PeerConnection and friends r=jesup,smaug,ekr
2013-01-17 17:11:14 -06:00
ccd851b350
Strip a= from trickle ICE candidates
...
Bug 829757 - Accept a=candidate ICE candidates (for now). r=ehugg
2013-01-17 09:18:06 -08:00
d1649a65c9
Bug 829461: Check for NULL media stream r=ehugg
2013-01-17 15:32:16 -06:00
51ac10a00c
Bug 826807: Clean up JSAPI error handling in PeerConnection constraints r=bz
2013-01-17 16:23:44 -05:00
3aa82fb6d0
Bug 837161: don't assert mainthread in AudioConduit create/init/destroy if in a unittest r=ehugg
2013-02-01 13:28:38 -05:00
b93e607197
Bug 822197 Handle creation of more PeerConnections than MAX_LINES r=jesup
2013-01-15 13:11:47 -08:00
e1e12091ba
Backed out changeset cebb008a72f9 (bug 824919) for mochitest b-c orange.
...
CLOSED TREE
2013-01-14 22:01:12 -05:00
f6c5f21152
Bug 824919: Weaken key references to PeerConnection and friends,r=jesup,smaug,ekr
2013-01-14 17:00:20 -06:00
a243ffc4dc
Bug 829761: Remove inactive check for sdpmode if port != 0, r=ekr
2013-01-14 16:38:38 -06:00
143985920e
Bug 828027 - Lower-case digest algorithm names r=ehugg
2013-01-14 09:10:38 -08:00
61b4d90206
Bug 829344 - Part 3: Fix gcc -Wlogical-op warnings in SIP code. r=jesup
2013-01-10 15:19:26 -08:00
cf8fbf13e1
Bug 827537. Refactor AudioChunk to support having separate buffers for each channel. r=jesup
...
--HG--
extra : rebase_source : 0aa26e1c3181d9fe5158520d4b33248bae0fa5d0
2012-11-22 18:04:27 +13:00
9bb826d9ee
Bug 785918 - Part 1: Replace PR_ARRAY_SIZE() with mozilla::ArrayLength() and MOZ_ARRAY_LENGTH(). r=ehsan
2013-01-05 23:37:25 -08:00
58cf683f4d
Bug 827359 - Disable unconditional use of NEON code in WebRTC DSP code. r=jesup
2013-01-08 18:30:10 +01:00
304e36fea0
Bug 579517 follow-up: Remove NSPR types that crept in
2013-01-07 18:21:50 -05:00
a9eb151cf8
Backout c9831bed6bb7 - Bug 818843 - This change broke building B2G on the mac. r=me
...
--HG--
extra : rebase_source : 7bef8dc1c394d45302f17379ab2cbd90ecda6975
2013-01-07 08:48:24 -08:00
fb7916fef7
Bug 818843 - Media changes. r=rjesup
2013-01-06 22:24:14 -05:00
221f6ac665
Bug 805251 - Minimal fix for creation refcount and transport release r=ekr,derf
2013-01-06 22:01:23 -05:00
86f510779d
Bug 825785 Signaling - monitor threads for shutdown r=jesup
2013-01-03 19:12:28 -08:00
41ca9a78eb
Bug 825611 - Fix unit tests to match fix to MediaPipeline. r=jesup
2013-01-05 14:52:11 -08:00
d9f2a41f28
Bug 825611 - Have MediaPipeline deliver as much media as requested. r=jesup
2013-01-05 09:02:36 -08:00
80ca76c7ad
Bug 820538: Added media-ptr check because of NS_DISPATCH_SYNC. r=rjesup
2013-01-04 23:01:56 -05:00
ec25adc6cd
bug 822956 - correct chunk_remaining calculation r=ekr
2013-01-04 14:41:25 -05:00
f9e86de52b
Bug 822956. ProcessAudioChunk needs to take account of AudioChunk::mOffset. r=jesup
2013-01-04 12:16:32 -05:00
3ae42c668d
bug 822956: Handle audio chunks that aren't 10ms in duration r=ekr
2013-01-03 18:37:55 -05:00
bec983b2ee
Bug 826529 - Increase default video bitrate. r=jesup
2013-01-03 19:17:42 -08:00
3ce45c1705
Bug 826008: Fix type tests for constraints objects r=smaug
2013-01-03 11:58:34 -05:00
0df7ae1002
Bug 825086: Removing errant free of body parts when encoding fails r=ekr
2013-01-02 15:06:12 -06:00
97291afe75
Bug 825785 cprDestroyThread on Windows should kill thread immediately r=jesup
2013-01-02 17:08:48 -08:00
8ba2e8f1c3
Bug 825439: Disconnect pipeline on STS Thread r=jesup
2012-12-29 09:24:34 -08:00
30b1a7cbaf
Bug 824263 - Shutdown: PeerConnectionMedia disconnect_all() + peer_ctx cleanup. r=jesup,ekr
2012-12-27 17:38:45 -05:00
8d4b69e76b
Bug 825526: Improve lifetime control of SourceMediaStream in gUM r=anant
2012-12-31 18:12:15 -05:00
5d015647a6
Bug 821071: Initialize all out parameters in VcmSIPCCBinding.cpp, r=ekr
2012-12-31 11:43:22 -06:00
6a7efc1149
Bug 825514: Add safety check to PeerConnectionCtx shutdown r=ekr
2012-12-31 12:34:44 -05:00
18af471013
Bug 824956: Activate media type when set to receive media by constraint r=jesup
2012-12-28 16:40:51 -06:00
d168fe854b
Bug 825336: Add constraint to suppress data channel r=jesup
2012-12-28 12:09:58 -08:00
fa8c05f7fa
Bug 825477 - Ignore return value from async calls in MediaPipeline. r=jesup
2012-12-30 07:53:48 -08:00
65602dce5c
Bug 825106: Use the correct pointer comparison when determining master r=jesup
2012-12-28 15:12:47 -06:00
f4e2d96791
Bug 820011: Shut down webrtc signling service on XPCOM shutdown r=derf
2012-12-28 17:04:32 -05:00
1ec34c7bd6
Bug 824220: Pass constraints through directly through message, deallocate when done r=jesup
2012-12-27 14:28:11 -06:00
cf9a82a2bc
Bug 815916 - Unbreak building signaling tests for system jpeg/pixman. r=rjesup
2012-12-26 10:15:53 -05:00
8eac1dfb3c
Bug 824397 - Use NS_ENSURE_{TRUE,FALSE,SUCCESS}_VOID(foo) instead of NS_ENSURE_{TRUE,FALSE,SUCCESS}(foo, ). r=derf.
2012-12-23 20:45:57 -08:00
94a8960630
Bug 824351 - Make PeerConnection.Close not check state. r=ehugg
2012-12-23 17:03:26 -08:00
e0f74b0102
Bug 821003: Replace snprintf macro with static function r=jesup
2012-12-21 14:55:41 -06:00
ff272006a8
Bug 818714: Set media enabled to FALSE unless added using addStream, r=ehugg
2012-12-20 16:05:28 -06:00
e86ce73501
Bug 797534: Update PeerConnectionImpl to use SDP from SIPCC, r=ehugg
2012-12-19 09:25:51 -06:00
fc51b305da
Bug 822158: Use async dispatch of Ice(Gathering)Completed to unwind stack r=jesup,ekr
2012-12-21 15:21:15 -05:00
35b1d04204
Bug 819856, Re-enabled WebRTC voice engine code for Android, r=jesup
2012-12-21 14:03:33 -08:00
4ccc635ab9
Bug 820102 - Clean up MediaPipeline threading (re-land after fix). r=derf,jesup
2012-12-21 06:03:22 -08:00
690f70ba61
Bug 821812, fix Android WebRTC signalling code hang by ensuring writable tmpdir for domain sockets, r=jesup
2012-12-21 10:29:49 -08:00
020839f9bb
Backout 21409a401d75 (bug 821292),9587e39f9a50 (bug 820102) for conflicts and assertions respectively, on a CLOSED TREE
2012-12-21 16:15:01 +00:00
470d172869
Bug 820102 - Clean up MediaPipeline threading. r=derf,jesup
2012-12-21 06:03:22 -08:00
fd7db16574
Backout bug 818714 because it introduces a sigabort crash
...
--HG--
extra : rebase_source : 4307433661215827bc238f514fa66758e27366bc
2012-12-20 13:47:12 +01:00
0a06495cf6
Bug 817430: Set initial controlled/controlling values, based on inbound T/F. r=jesup
2012-12-02 13:49:56 -08:00
fb7884b67b
Bug 818714: Set media enabled to FALSE unless added using addStream, r=ehugg
2012-12-19 20:52:32 -06:00
fe0e2c5d82
Bug 794240 followup: Fixes implicit function warning that broke linkage + symmetric fixes r=jesup
2012-12-19 13:00:49 -05:00
9f08e63e8f
Bug 822704: Enable WEBRTC_TRACE() logging via NSPR_LOG_MODULES and rename signaling log module r=derf
2012-12-18 23:27:38 -05:00
10c0782671
Bug 794240: Disable timerthread + close sockets + cleanup tmp-files on shutdown + file permissions. r=jesup
2012-12-14 16:15:21 -05:00
64a2a740d6
Bug 806829 - Signaling - remove g_deviceInfo.name r=jesup
2012-12-14 13:43:06 -08:00
9443517959
bug 821299 - Make ALSA WebRTC backend expose proper IDs using device names. r=jesup
...
--HG--
extra : rebase_source : 9ab962a7de537d55140a566bff1df2b0794d63fa
2012-12-17 08:28:24 -05:00
b977feb178
Bug 807012 - Signaling - reorder initiialization of CallControlManagerImpl r=jesup
2012-12-17 11:01:32 -08:00
6b59cd00d7
Bug 821621 - Fix unused variable warning in PeerConnectionCtx.cpp. r=jesup
2012-12-15 00:22:47 -08:00
5e38742f08
Bug 806375: cleanup DataChannel, esp. channel close and connection shutdown r=mcmanus
2012-12-13 23:30:11 -05:00
010027f7be
backout 916f6915112d bug 806375 for anonymous enum bustage on linux
2012-12-14 00:26:58 -05:00
a92d3614dc
Bug 806375: cleanup DataChannel, esp. channel close and connection shutdown r=mcmanus
2012-12-13 23:30:11 -05:00
6b755ededd
Bug 816822 - Fix build files, unit tests, and clone some linux headers to make WebRTC unit tests build on Android, r=dmose,ted
2012-12-12 12:05:57 -08:00
d1407c8702
Bug 815883 - fix WebRTC builds for ARM chips with neon FPUs by cloning some linux headers and fixing build files. r=dmose,ted
2012-12-12 12:05:54 -08:00
c5f6539df3
Bug 820550 - Signaling - stop checking API state in CloseInt r=ekr
2012-12-12 09:39:51 -08:00
1fd24dc33b
Bug 820671 Make PCImpl a tombstone if mMedia is null r=jesup
2012-12-11 19:45:09 -08:00
596b7622b0
Bug 820559: Workaround old Android gcc build-bustage by casting anonymous enum template args to ints, r=ekr
2012-12-12 10:15:49 -08:00
30d7f1c6f4
bug 820769 - fix mozmake.py to generate correct common.mk include. r=jesup
2012-12-12 11:32:05 -05:00
fd7090483a
Bug 792175 - Move PeerConnection operations onto main thread. r=jesup,derf,ehugg
2012-09-26 10:14:23 -07:00
d5de987562
Bug 816575 - Remove call to SetRenderAndroidVM when not using internal renderer. r=tterribe
2012-12-11 20:11:15 +01:00
944f3ba006
Bug 813913 - WebRTC signaling code doesn't yet build on Android. r=dmose rs=dmose
2012-12-10 16:12:50 -08:00
f86cd2d8b1
Bug 752657: backout sigslot multithreaded due to deadlocks rs=jesup
2012-12-11 12:54:24 -05:00
9ae439bc32
Bug 752657: switch all uses of sigslot to default to multithreaded (win32 already defaults that way) r=ekr
2012-12-11 07:59:57 -05:00
412ada1719
Bug 806822: avoid race condition in gathering stats r=ehugg
2012-12-11 07:58:53 -05:00
0d339d0c65
Bug 819246 - arm_neon is meaningful only when armv7 is true. r=jesup
2012-12-10 09:05:00 -05:00
cf7579d105
Merge mozilla-central to mozilla-inbound
2012-12-10 14:03:11 +00:00
56d1df4c55
Bug 814693 - Remove __MIPSEL__ block from media/webrtc/trunk/src/typedefs.h, it's handled under __mips__. r=jesup,DONTBUILD
2012-12-10 09:09:27 +01:00
33daa6a518
Bug 816112: Configure webrtc video/RTCP parameters for better defaults r=jesup
2012-11-28 08:34:10 -08:00
4610ecbd72
Bug 819364 - Webrtc signaling unit test does not build on Qt port. r=rjesup
2012-12-09 09:28:37 -08:00
6de7befa37
Bug 818817 - Fix some compiler warnings, r=Ms2ger
2012-12-09 22:53:19 +05:30
8a57275446
Bug 817065: Replace vcm_media_payload_t with structure leveraging rtp_ptype constants r=ehugg,jesup
2012-12-06 11:36:43 -06:00
1bd5fe4a2b
Bug 817488 Audio Send and Recv support for PC tests r=ekr
2012-12-05 13:37:32 -08:00
9c465795b8
Bug 814693 - Allow webrtc to build on more architectures. r=jesup
...
--HG--
extra : rebase_source : 7d22643c1b4b944595bfe33ac8f5925af2b8bcbd
2012-12-01 09:55:48 +01:00
18ef53f5bc
Bug 814329 Protect fim_process_event from NULL input r=jesup
2012-11-28 08:38:50 -08:00
7732931b5f
Bug 750869 - Build system support for --enable-webrtc for Android (off by default). r=ted
2012-12-04 16:27:18 +01:00
d53c47bb99
Bug 817481 - Build failure on WebRTC unit test with --with-system-libvpx. r=jesup
2012-12-03 17:01:16 +09:00
aa9120552c
Bug 810220 - Patch to fix SDP Codec Negotiation Issues (revised) r=ekr,jesup
2012-11-26 02:36:43 -08:00
dfc4104a87
Bug 817451, Correctly set mRole in PC unit tests. r=ehugg
2012-12-02 18:06:21 -08:00
c4ec584655
Bug 813241: Update config/system-headers and make wrapping of NSPR & NSS headers more robust, r=glandium
...
--HG--
extra : rebase_source : 5ba0a83110268ff489df7b3e0a8a9219711247b1
2012-11-22 11:15:01 -08:00
fcaf160cf1
Bug 812636: additional checks for failure for a Conduit be created r=ehugg
2012-11-28 03:28:30 -05:00
f00177debd
Bug 815928 part 2: Fix typo s/elment/elment/ in comments & tests. DONTBUILD, rs=Waldo
2012-11-27 19:15:36 -08:00
79078b7ce8
Bug 815928 part 1: Fix typo s/elelment/element/ in comments & tests. DONTBUILD, rs=Waldo
2012-11-27 19:15:35 -08:00
e69205c689
Bug 814734 - Fixed Log format string mismatches r=ekr
2012-11-26 08:38:14 -08:00
4152b3d262
Bug 805662, Crash in mediaconduit_unittest. r=jesup
2012-11-22 10:57:37 -08:00
17299e21c3
Bug 814038: Fixing codec negotiation to use actual payload in SDP rather than preferred payload when populating codec-specific parameters r=ehugg
2012-11-21 15:20:12 -06:00
c0c2b72765
Bug 813746. Link the webrtc unit tests to ZLIB as needed. r=bz
2012-11-21 02:16:03 -05:00
34aa729ec6
Bug 813723 Reorder state assignment in unit tests r=ekr
2012-11-20 13:47:07 -08:00
0aff34f3d3
Bug 811118 build webrtc unittests by default but run only some r=jesup
2012-11-12 15:34:05 -08:00
8980d5151e
Bug 813212 Update signaling unittests to match latest IDL r=jesup
2012-11-19 09:59:36 -08:00
c92c0012eb
Bug 799419: Force NSS startup during PeerConnection Initialize r=bsmith,jesup
2012-11-18 07:43:26 -08:00
8e428d4084
Bug 812886: Watch network (tear)down events and kill PeerConnections r=ekr,smaug,bsmith
2012-11-18 23:53:14 -05:00
67fb41348c
Bug 812641: Shut down SipCC instance when number of PeerConnections == 0 (reland) r=ehugg
2012-11-17 23:03:58 -05:00
41197e27e3
Bug 812641: Shut down SipCC instance when number of PeerConnections==0 r=ehugg
2012-11-18 01:42:40 -05:00
f9cd0623b0
Backed out changeset b527670b6728 (bug 811118) for orange.
2012-11-16 21:24:14 -05:00
0526baae8c
Bug 810544 - don't include <iostream> in webrtc code where it's not necessary; r=jesup
2012-11-09 11:08:22 -05:00
ae04e69977
Bug 811118 build webrtc unittests by default r=jesup
2012-11-12 15:34:05 -08:00
5748e6d4d9
Bug 806830: Enforce initializing strlib before using r=ehugg
2012-11-16 14:37:08 -05:00
8523b3bb2f
Bug 811183: Recursive GC In PeerConnection shutdown; r=jesup
2012-11-16 10:27:30 -08:00
20be5c457f
Bug 810363 - Reject non-stereo, non-48 kHz Opus streams, r=ehugg
2012-11-15 15:09:39 -08:00
6394681a81
Bug 810353 - Offer Opus as stereo instead of mono, r=ehugg
2012-11-15 15:09:39 -08:00
decae3d858
Bug 811695: disable internal socket transports for getUserMedia Audio capture r=derf
2012-11-15 17:58:40 -05:00
9a7aea6d5b
Bug 803318: Improved handling of constraints and more tests. r=ekr
2012-11-14 11:25:14 -06:00
bb4830f48c
Bug 811333 Signaling - Fix size of thread ID for Windows build r=jesup
2012-11-13 09:23:04 -08:00
f0e9345907
Bug 806407 Fix type in struct passed into msgsnd f=jesup
2012-11-09 08:47:00 -08:00
22d38f2013
Bug 809950 - fix webrtc signaling Wrapper code to not introduce static initializers; r=jesup
2012-11-08 13:45:39 -05:00
0ce97e7c2d
Bug 807929: Make DataChannel refcounted r=mcmanus
2012-11-02 15:28:13 -04:00
988e5fc796
Bug 807259: Fix shutdown for WebRTC standalone unit tests. r=anant
2012-10-31 16:31:47 +01:00
59cd59057f
Bug 805279: make sure we pass values for formatted log messages r=derf
2012-10-30 15:31:09 -04:00
2f2c416342
Bug 806335: MediaPipeline destroys TransportFlow on wrong thread; r=anant
2012-10-29 15:35:12 +01:00
a720cd09bd
Bug 806306: Fix compile bustage for MediaConduit; r=anant
2012-10-29 15:33:19 +01:00
b1feab9a84
Bug 805703. Part 1: MediaStreamGraph::CreateInputStream -> CreateSourceStream. r=jesup
...
--HG--
extra : rebase_source : 3c327e9ead92508f12df4b95f2fd24fa2ba97ab5
2012-10-29 17:36:31 +13:00
39a0bbea1d
Bug 805701: protect against NULL fcb pointer r=ehugg
2012-10-27 21:47:39 -04:00
6ee946c418
Bug 801221: Move mtransport operations onto STS thread; r=jesup
2012-10-18 13:01:52 -07:00
5686d907c0
Bug 805533 Send SDP parse errors through single function for reporting r=jesup
2012-10-26 13:05:50 -07:00
acc22ea0d6
Bug 805254. Part 8: Consolidate audio sample processing code using templates over the format types. r=kinetik
...
Replace nsAudioStream::Format with an AUDIO_OUTPUT_FORMAT enum value so we
can use it as a template parameter.
Introduce AudioSampleTraits<AudioSampleFormat> to give us access to the C++ type
corresponding to an enum value.
Move SampleToFloat/FloatToSample to AudioSampleFormat.h.
Introduce ConvertAudioSamples and ConvertAudioSamplesWithScale functions
and use them from various places.
Moves AudioDataValue to AudioSampleFormat.h. The name isn't great, but it'll do.
2012-10-25 23:09:40 +13:00
eb84abab22
Bug 805254. Part 7: Move SampleFormat to mozilla::AudioSampleFormat in its own file. r=kinetik
2012-10-25 23:09:40 +13:00
9bfc64887e
Bug 805254. Part 4: Remove FORMAT_U8 from nsAudioStream::SampleFormat. r=kinetik
...
We also give nsWaveReader its own separate format enum.
2012-10-25 23:09:39 +13:00
4f8fe31d2a
Merge the last PGO-green inbound chnageset to m-c.
2012-10-25 21:14:50 -04:00
4b44447e10
Bug 803842: Fix incorrect index for warning length; bullet-proof the code some r=derf
2012-10-24 15:39:55 -04:00
e2e9343487
Bug 803744 - Allow signaling startup when already started r=jesup
2012-10-24 15:46:40 -07:00
f90c9929ee
Bug 802694: Pass along constraints from PC JS module to PCImpl; r=ekr,jesup
2012-10-25 12:24:30 -07:00
8e702f60c5
Backed out 2 changesets (bug 579517)
...
Backed out changeset 5298adc70963
Backed out changeset 86ccf7c918ce (bug 579517)
2012-10-25 12:32:24 -04:00
f3c0ac7a23
Code hygiene: don't use PR_TRUE and PR_FALSE, and use stdint types instead of PRInt types (no bug really, but you could say bug 579517)
2012-10-25 11:48:19 -04:00
c093eaa46e
Bug 803881: Fix wrong number of samples for webrtc media unit tests r=ekr
2012-10-24 16:54:33 -04:00
d241ae5621
Bug 729541 fix for syntax error in signaling_unittests r=jesup
2012-10-23 21:26:27 -07:00
4a3a65feec
Bug 797512 Signaling: simplify set_dtls_fingerprint r=jesup
2012-10-22 16:38:52 -07:00
31407fee99
Bug 803875 MediaConstraints - fix length handling r=jesup
2012-10-22 13:25:44 -07:00
582c09cab8
Bug 729541: Remove 'offer' parameter from RTCPeerConnection.createAnswer r=jesup
2012-10-15 16:50:10 +01:00
76f9897831
Bug 803086: Process NULL image chunks in NotifyQueuedTrackChanges() r=ekr
2012-10-19 10:58:06 -04:00
822bfa909d
Bug 800688: remove local_dynamic_payload_type_value from negotiation r=ehugg,jesup
2012-10-17 15:57:52 +01:00
6726f96604
Bug 798873 Patch 3 - flex_string fix for Windows vsnprintf r=jesup
2012-10-17 18:57:57 -07:00
2939d5a843
Bug 799975 - Fix webrtc when building with system nspr/nss. r=ted
2012-10-17 16:33:23 +02:00
ca37c0beaf
Bug 798926 - Define INTEGER_TYPES_H and others for srtp.h. r=jesup
2012-10-17 16:28:33 +02:00
b53102d6ef
Bug 786152: Increment DataChannel SCTP port in a WebRTC JSEP answer r=jesup
2012-09-28 11:36:02 +01:00
b2af5ec3d2
Bug 784515: add hints to webrtc/signaling CreateOffer and CreateAnswer APIs r=jesup
2012-09-28 11:09:09 +01:00
767c08f204
Fix for bug 711628 (Implement PeerConnection.localStreams/remoteStreams). r=bz.
...
--HG--
extra : rebase_source : e21a429d85fc60a972752a2a7deb88a7cb648f3c
2012-09-13 18:04:31 +02:00
31d9b62d0d
Bug 800847: Correctly regenerate Makefiles from gyp files, and handle fancy symlinked objdirs r=ted
2012-10-12 16:06:33 -04:00
4c76215d4f
Bug 800611 - Trailing whitepace removed from signaling code r=jesup
2012-10-12 08:15:24 -07:00
478141e4d2
Bug 796457 - upgrade license to MPL 2.
2012-10-12 15:58:11 +01:00
f214df21b4
Backout 3044539fec87 (bug 799465),2dbcd6d16b43 (bug 798264), f7019f73a5f0 (bug 711628), a484a3a904da, 7154061ddc00 & 7e7fc42021c1 (bug 799465) for burning
2012-10-12 14:45:38 +01:00
0958866530
Fix for bug 711628 (Implement PeerConnection.localStreams/remoteStreams). r=bz.
...
--HG--
extra : rebase_source : d7be35954ba69563ed26b2173c75fe9e74707847
2012-09-13 18:04:31 +02:00
5ddad9093f
Bug 800343 - Cast nsresult to uint32_t to output to streams; r=ehsan
2012-10-11 14:48:31 +02:00
296f7d04db
Bug 792811 - Check creation of WebRTC transportflows r=jesup
2012-10-09 12:29:01 -07:00
9d52da089c
Bug 800502 - Signaling - Removed unused code that required XML handling r=jesup
2012-10-11 14:07:50 -07:00
e62c39b8e5
Bug 800558 - Signaling - return proper error code in fsmdef.c r=jesup
2012-10-11 14:32:45 -07:00
25ac2d3e43
Bug 798873 - Signaling SDP construction uses flex_string r=jesup
2012-10-08 16:43:33 -07:00
96fa8a04e3
Bug 800546 - Ensure GYP-generated make files depend on the proper file; r=ted
...
Previously they could depend on mozmake.pyc. Now, we ensure they
reference the original .py file.
2012-10-11 15:21:19 -07:00
c2c0b53beb
Bug 800401 - Make sure that the WebRTC build system does not impose the -Werror flag on all Linux/Mac builds; r=jesup
...
In the future, we need to hook these up to the FAIL_ON_WARNINGS machinery.
2012-10-11 13:42:12 -04:00
1d84db82f6
Backout d6e84a64d3d4 (bug 798969) for burning
2012-10-11 14:37:44 +01:00
5020db2e03
Bug 798969. Handle symlinked srcdirs by avoiding use of Python's __file__. r=jesup
2012-10-08 17:45:10 +13:00
f7560ac779
Bug 791278: Protect PeerConnection setLocal/RemoteDescription from NULL input r=jesup
2012-10-09 15:14:51 -07:00
c601df4d52
Bug 791270 - Protect AddStream from NULL input and cause a JS error to be thrown r=jesup
2012-10-09 14:03:13 -07:00
9151531465
Bug 799477: missing stdarg.h in CSFLog.h (typically gcc 4.4.x) r=ehugg
2012-10-09 14:46:34 -04:00
1188de1f60
Bug 799246: Conditionally enable webrtc unit tests r=jesup,ted,cjones
2012-10-08 18:56:00 -07:00
3e431eac40
Bug 799071: clean up clang warnings in media/webrtc/signaling (-Werror) r=ehugg
2012-10-08 17:11:10 -04:00
9236c6f2f9
Bug 798873 - Signaling - increase max SDP size (temporary until better fix). r=jesup,ekr
2012-10-07 22:28:52 -07:00
5a3edf0816
BUg 799076: Add temporary error definitions to media/webrtc/signaling for VC9 and below r=derf
2012-10-08 11:08:52 -04:00
de872c99fe
Bug 798892: Wrong enum type in gCCApp.state, breaks bleeding-edge clang builds r=ehugg
2012-10-07 21:55:45 -04:00
058aa73640
Bug 579517 follow-up: Remove NSPR types that crept in
2012-10-07 18:26:08 -04:00
be9228b287
Bug 798825: Add DataChannel DOM interfaces to RTCPeerConnection; r=smaug
2012-10-07 01:34:30 -04:00
e61ba69837
Bug 694807: Implement PeerConnection C++ module; r=jst,jesup,ekr
2012-10-07 01:34:30 -04:00
7e1291fe34
Bug 790517: mtransport - Generic media transport subsystem for ICE and DTLS r=jesup,bsmith,mcmanus
2012-10-02 13:04:58 -07:00
52df55c57b
Bug 792188: Fixes to signaling MediaPipeline and Srtpflow r=jesup,derf
2012-10-07 01:34:29 -04:00
d7261ed122
Bug 792188: MediaPipeline changes for YUV buffer handling, put ImageContainer back to normal r=ekr
2012-10-04 05:35:26 -04:00
f769554f67
Bug 792188: rollup of changes to signaling from alder r=jesup,ekr,derf,ehugg,ted
...
Bug 797258: Increase max sdp line length to work with longer SHA-256 fingerprints. r=jesup
2012-10-07 01:34:29 -04:00
1995a1f5d1
Bug 792188: Import SIPCC into media/webrtc/signaling rs=jesup r=gerv
2012-03-13 11:22:14 -07:00
61f2eb940e
Bug 797671: cleanup from importing webrtc.org update r=ted,glandium (Part is bug 778801 r=derf)
2012-10-04 12:09:35 -04:00
de7aef9986
Bug 797671: Import Webrtc.org code from stable branch 3.12 (rev 2820) rs=jesup
...
--HG--
rename : media/webrtc/trunk/tools/valgrind/tsan/OWNERS => media/webrtc/trunk/build/OWNERS
rename : media/webrtc/trunk/build/android/chrome_test_server_spawner.py => media/webrtc/trunk/build/android/pylib/chrome_test_server_spawner.py
rename : media/webrtc/trunk/build/android/cmd_helper.py => media/webrtc/trunk/build/android/pylib/cmd_helper.py
rename : media/webrtc/trunk/build/android/test_package_executable.py => media/webrtc/trunk/build/android/pylib/test_package_executable.py
rename : media/webrtc/trunk/build/android/valgrind_tools.py => media/webrtc/trunk/build/android/pylib/valgrind_tools.py
rename : media/webrtc/trunk/src/common_audio/signal_processing/spl_sqrt_floor.s => media/webrtc/trunk/src/common_audio/signal_processing/spl_sqrt_floor_arm.s
rename : media/webrtc/trunk/src/common_video/libyuv/include/libyuv.h => media/webrtc/trunk/src/common_video/libyuv/include/webrtc_libyuv.h
rename : media/webrtc/trunk/src/modules/rtp_rtcp/test/test_bwe/unit_test.cc => media/webrtc/trunk/src/modules/remote_bitrate_estimator/bitrate_estimator_unittest.cc
rename : media/webrtc/trunk/src/modules/rtp_rtcp/source/bwe_defines.h => media/webrtc/trunk/src/modules/remote_bitrate_estimator/include/bwe_defines.h
rename : media/webrtc/trunk/src/modules/rtp_rtcp/source/remote_rate_control.cc => media/webrtc/trunk/src/modules/remote_bitrate_estimator/remote_rate_control.cc
rename : media/webrtc/trunk/src/modules/rtp_rtcp/source/remote_rate_control.h => media/webrtc/trunk/src/modules/remote_bitrate_estimator/remote_rate_control.h
rename : media/webrtc/trunk/src/modules/rtp_rtcp/source/rtcp_sender_test.cc => media/webrtc/trunk/src/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
rename : media/webrtc/trunk/src/modules/rtp_rtcp/source/rtp_header_extension_test.cc => media/webrtc/trunk/src/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
rename : media/webrtc/trunk/src/modules/rtp_rtcp/source/rtp_packet_history_test.cc => media/webrtc/trunk/src/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
rename : media/webrtc/trunk/src/modules/rtp_rtcp/source/rtp_sender_test.cc => media/webrtc/trunk/src/modules/rtp_rtcp/source/rtp_sender_unittest.cc
rename : media/webrtc/trunk/src/modules/rtp_rtcp/source/rtp_utility_test.cc => media/webrtc/trunk/src/modules/rtp_rtcp/source/rtp_utility_unittest.cc
rename : media/webrtc/trunk/src/modules/rtp_rtcp/source/transmission_bucket_test.cc => media/webrtc/trunk/src/modules/rtp_rtcp/source/transmission_bucket_unittest.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/automated_mode.cc => media/webrtc/trunk/src/modules/video_capture/main/test/video_capture_main_mac.mm
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/source/Android.mk => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/Android.mk
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8.h => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/include/vp8.h
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/include/vp8_common_types.h
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/reference_picture_selection.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.h => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/reference_picture_selection.h
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection_unittest.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/reference_picture_selection_unittest.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/temporal_layers.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.h => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/temporal_layers.h
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers_unittest.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/temporal_layers_unittest.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/benchmark.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.h => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/benchmark.h
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/dual_decoder_test.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.h => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/dual_decoder_test.h
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/normal_async_test.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.h => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/normal_async_test.h
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/packet_loss_test.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.h => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/packet_loss_test.h
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/rps_test.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.h => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/rps_test.h
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/tester.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/tester.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/test/vp8_unittest.cc
rename : media/webrtc/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.cc => media/webrtc/trunk/src/modules/video_coding/codecs/vp8/vp8.cc
rename : media/webrtc/trunk/test/OWNERS => media/webrtc/trunk/src/test/OWNERS
rename : media/webrtc/trunk/src/video_engine/test/libvietest/helpers/bit_flip_encryption.cc => media/webrtc/trunk/src/test/libtest/helpers/bit_flip_encryption.cc
rename : media/webrtc/trunk/src/video_engine/test/libvietest/helpers/random_encryption.cc => media/webrtc/trunk/src/test/libtest/helpers/random_encryption.cc
rename : media/webrtc/trunk/src/video_engine/test/libvietest/include/bit_flip_encryption.h => media/webrtc/trunk/src/test/libtest/include/bit_flip_encryption.h
rename : media/webrtc/trunk/src/video_engine/test/libvietest/include/random_encryption.h => media/webrtc/trunk/src/test/libtest/include/random_encryption.h
rename : media/webrtc/trunk/test/metrics.gyp => media/webrtc/trunk/src/test/metrics.gyp
rename : media/webrtc/trunk/test/run_all_unittests.cc => media/webrtc/trunk/src/test/run_all_unittests.cc
rename : media/webrtc/trunk/test/test.gyp => media/webrtc/trunk/src/test/test.gyp
rename : media/webrtc/trunk/test/test_suite.cc => media/webrtc/trunk/src/test/test_suite.cc
rename : media/webrtc/trunk/test/test_suite.h => media/webrtc/trunk/src/test/test_suite.h
rename : media/webrtc/trunk/test/testsupport/fileutils.cc => media/webrtc/trunk/src/test/testsupport/fileutils.cc
rename : media/webrtc/trunk/test/testsupport/fileutils.h => media/webrtc/trunk/src/test/testsupport/fileutils.h
rename : media/webrtc/trunk/test/testsupport/fileutils_unittest.cc => media/webrtc/trunk/src/test/testsupport/fileutils_unittest.cc
rename : media/webrtc/trunk/test/testsupport/frame_reader.cc => media/webrtc/trunk/src/test/testsupport/frame_reader.cc
rename : media/webrtc/trunk/test/testsupport/frame_reader.h => media/webrtc/trunk/src/test/testsupport/frame_reader.h
rename : media/webrtc/trunk/test/testsupport/frame_reader_unittest.cc => media/webrtc/trunk/src/test/testsupport/frame_reader_unittest.cc
rename : media/webrtc/trunk/test/testsupport/frame_writer.cc => media/webrtc/trunk/src/test/testsupport/frame_writer.cc
rename : media/webrtc/trunk/test/testsupport/frame_writer.h => media/webrtc/trunk/src/test/testsupport/frame_writer.h
rename : media/webrtc/trunk/test/testsupport/frame_writer_unittest.cc => media/webrtc/trunk/src/test/testsupport/frame_writer_unittest.cc
rename : media/webrtc/trunk/test/testsupport/gtest_prod_util.h => media/webrtc/trunk/src/test/testsupport/gtest_prod_util.h
rename : media/webrtc/trunk/test/testsupport/metrics/video_metrics.cc => media/webrtc/trunk/src/test/testsupport/metrics/video_metrics.cc
rename : media/webrtc/trunk/test/testsupport/metrics/video_metrics.h => media/webrtc/trunk/src/test/testsupport/metrics/video_metrics.h
rename : media/webrtc/trunk/test/testsupport/metrics/video_metrics_unittest.cc => media/webrtc/trunk/src/test/testsupport/metrics/video_metrics_unittest.cc
rename : media/webrtc/trunk/test/testsupport/mock/mock_frame_reader.h => media/webrtc/trunk/src/test/testsupport/mock/mock_frame_reader.h
rename : media/webrtc/trunk/test/testsupport/mock/mock_frame_writer.h => media/webrtc/trunk/src/test/testsupport/mock/mock_frame_writer.h
rename : media/webrtc/trunk/test/testsupport/packet_reader.cc => media/webrtc/trunk/src/test/testsupport/packet_reader.cc
rename : media/webrtc/trunk/test/testsupport/packet_reader.h => media/webrtc/trunk/src/test/testsupport/packet_reader.h
rename : media/webrtc/trunk/test/testsupport/packet_reader_unittest.cc => media/webrtc/trunk/src/test/testsupport/packet_reader_unittest.cc
rename : media/webrtc/trunk/test/testsupport/unittest_utils.h => media/webrtc/trunk/src/test/testsupport/unittest_utils.h
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rename : media/webrtc/trunk/src/voice_engine/main/source/voe_hardware_impl.cc => media/webrtc/trunk/src/voice_engine/voe_hardware_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_hardware_impl.h => media/webrtc/trunk/src/voice_engine/voe_hardware_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.cc => media/webrtc/trunk/src/voice_engine/voe_neteq_stats_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.h => media/webrtc/trunk/src/voice_engine/voe_neteq_stats_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_network_impl.cc => media/webrtc/trunk/src/voice_engine/voe_network_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_network_impl.h => media/webrtc/trunk/src/voice_engine/voe_network_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc => media/webrtc/trunk/src/voice_engine/voe_rtp_rtcp_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.h => media/webrtc/trunk/src/voice_engine/voe_rtp_rtcp_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_video_sync_impl.cc => media/webrtc/trunk/src/voice_engine/voe_video_sync_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_video_sync_impl.h => media/webrtc/trunk/src/voice_engine/voe_video_sync_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_volume_control_impl.cc => media/webrtc/trunk/src/voice_engine/voe_volume_control_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_volume_control_impl.h => media/webrtc/trunk/src/voice_engine/voe_volume_control_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voice_engine_defines.h => media/webrtc/trunk/src/voice_engine/voice_engine_defines.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voice_engine_impl.h => media/webrtc/trunk/src/voice_engine/voice_engine_impl.h
rename : media/webrtc/trunk/testing/gtest/COPYING => media/webrtc/trunk/testing/gtest/LICENSE
rename : media/webrtc/trunk/tools/gyp/test/same-name/gyptest-all.py => media/webrtc/trunk/tools/gyp/test/same-source-file-name/gyptest-all.py
rename : media/webrtc/trunk/tools/gyp/test/same-name/gyptest-default.py => media/webrtc/trunk/tools/gyp/test/same-source-file-name/gyptest-default.py
rename : media/webrtc/trunk/tools/gyp/test/same-name/src/all.gyp => media/webrtc/trunk/tools/gyp/test/same-source-file-name/src/all.gyp
rename : media/webrtc/trunk/tools/gyp/test/same-name/src/func.c => media/webrtc/trunk/tools/gyp/test/same-source-file-name/src/func.c
rename : media/webrtc/trunk/tools/gyp/test/same-name/src/prog1.c => media/webrtc/trunk/tools/gyp/test/same-source-file-name/src/prog1.c
rename : media/webrtc/trunk/tools/gyp/test/same-name/src/prog2.c => media/webrtc/trunk/tools/gyp/test/same-source-file-name/src/prog2.c
rename : media/webrtc/trunk/tools/gyp/test/same-name/src/subdir1/func.c => media/webrtc/trunk/tools/gyp/test/same-source-file-name/src/subdir1/func.c
rename : media/webrtc/trunk/tools/gyp/test/same-name/src/subdir2/func.c => media/webrtc/trunk/tools/gyp/test/same-source-file-name/src/subdir2/func.c
rename : media/webrtc/trunk/tools/gyp/test/msvs/precompiled/hello.c => media/webrtc/trunk/tools/gyp/test/win/precompiled/hello.c
rename : media/webrtc/trunk/tools/gyp/test/msvs/precompiled/hello2.c => media/webrtc/trunk/tools/gyp/test/win/precompiled/hello2.c
rename : media/webrtc/trunk/tools/gyp/test/msvs/precompiled/precomp.c => media/webrtc/trunk/tools/gyp/test/win/precompiled/precomp.c
2012-10-04 12:09:31 -04:00
35bc9a72e8
Bug 785542 - Convert usages of PR_MIN and PR_MAX to NS_MIN and NS_MAX; r=ehsan
...
Occurences of PR_MAX in layout/style/nsCSSProps.cpp and xpcom/glue/nsTArray.h
can not be converted without C++11 support (constexpr).
--HG--
extra : rebase_source : 3b4f7e26690fad487dd11594449948411d4e79bc
2012-09-27 23:44:47 +02:00
06e7cc23e4
Bug 794297: Block the android build system from looking in mozilla-central. r=mwu
...
--HG--
rename : media/webrtc/Android.mk => Android.mk
2012-09-25 17:04:01 -07:00
e734d6f1a2
Bug 792343 - Enable FAIL_ON_WARNINGS in some more of dom/ (second batch); r=mounir
2012-09-20 09:55:36 +02:00
70d7c821af
Bug 774032 bonus - Use @DEPTH@ and @relativesrcdir@ in Makefile.in. r=ted
2012-08-04 20:26:44 +02:00
ca978c44cc
Bug 774060 - Compiling Firefox fails --with-system-libvpx when using libvpx 1.1 or later. r=tterribe
2012-07-24 11:57:47 +09:00
117e10ca3f
Bug 694817: Turn off webrtc's internal video renderer and protobuf (take 2) r=ted
2012-07-23 22:35:35 -04:00
90f10ba266
bug 773454. Pass "-D_VARIADIC_MAX=10" to the compiler on Windows when building gtest. r=ted
2012-07-23 18:07:58 -07:00
c7a1d5c2d8
Bug 772201: remove relative topsrcdir/srcdir/etc paths from gyp-sourced Makefiles - fixes symlinked objdirs for linux/mac r=ted
2012-07-12 18:14:14 -04:00
79c797a5e0
Bug 771981: Don't build webrtc except for Linux/Mac/Windows; don't check for chrome dlls on windows r=bsmedberg
2012-07-09 14:34:33 -04:00
c439649aed
Bug 770230: Remove broken symlinks in webrtc to third_party/google-visualization-python DONTBUILD r=derf
2012-07-06 12:15:49 -04:00
dd8ac25756
Bug 767250: Stop B2G/Gonk from trying to build Android.mk's in media/webrtc rs=mwu
2012-06-22 19:07:41 -04:00
b3b0fc1b72
Bug 749889 and Bug 688178: Make webrtc build without referencing third_party modules not in first tranche r=ted
2012-06-20 07:27:50 -04:00
0edea6e242
Bug 757637: Rollup makesystem changes for webrtc r=khuey r=ted f=glandium
2012-06-20 07:27:43 -04:00
9ccdfd5b08
Bug 766253: Fix type for kARGBToV table (upstream issue 188 at webrtc.org) r=derf
2012-06-20 07:27:32 -04:00
7ce436aea4
Bug 757637: Rollup media/webrtc/trunk changes from webrtc.org drop r=ted r=derf
2012-06-20 07:27:22 -04:00
ec2abb1282
bug 731407: need to include <assert.h> in non-debug builds r=cpearce
2012-04-04 14:49:12 -04:00
ebbb536539
bug 731407: Remove DShow BaseClass usage from webrtc drop r=cpearce
2012-03-15 23:06:35 +00:00
a9df4a23d9
Bug 749889: Webrtc import of rev 2047, with most of third_party and test/data removed rs=ted
2012-06-21 07:34:58 -04:00
Alexandros Chronopoulos
Randell Jesup
Ehsan Akhgari
Adam Roach [:abr]
Robert O'Callahan
Andrew Miller
Ryan VanderMeulen
Daniel Holbert
Ed Morley
Boris Zbarsky
Phil Ringnalda
Ethan Hugg
EKR
Justin Lebar
Jan-Ivar Bruaroey
Steven Lee
Nicholas Nethercote
Chris Peterson
Gian-Carlo Pascutto
Doug Turner
Maire Reavy
Jan Beich
Dan Mosedale
Henrik Skupin
Ted Mielczarek
Mike Hommey
Oleg Romashin
Saurabh Anand
Suhas Nandakumar
Makoto Kato
Brian Smith
Nathan Froyd
Timothy B. Terriberry
Anant Narayanan
Enda Mannion
Peter Van der Beken
Gervase Markham
Aryeh Gregor
Gregory Szorc
foudfou
Chris Jones
Ms2ger
Andrew Benton
Tim Abraldes
Bas Schouten